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SPA2102-AN - Single Port Router
Administration Manual
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Cisco SPA2102-AN - Single Port Router, SPA2102, SPA3102, SPA8000, SPA8800 Administration Manual

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Cisco Small Business SPA2102, SPA3102, SPA8000, SPA8800, PAP2T
Analog Telephone Adapters
ADMINISTRATION
GUIDE
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© 2009 Cisco Systems, Inc. All rights reserved. OL-17901-04
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Contents
About This Document 11
Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters 16
Comparison of ATA Devices 17
ATA Connectivity Requirements 20
PAP2T Connectivity 21
SPA2102 Connectivity 22
SPA3102 Connectivity 23
SPA8000 Connectivity 24
SPA8800 Connectivity 25
ATA Software Features 26
Voice Supported Codecs 26
SIP Proxy Redundancy 28
Other ATA Software Features 28
Chapter 2: Basic Administration and Configuration 36
Basic Services and Equipment Required 36
Downloading Firmware 37
Basic Installation and Configuration 37
Upgrading the Firmware for the ATA Device 37
Setting up Your ATA Device 38
Using the Administration Web Server 39
Connecting to the Administration Web Server 40
Setting Up the WAN Configuration for Your ATA Device 40
Registering to the Service Provider 42
Advanced Configurations 43
Upgrading, Rebooting, and Resyncing Your ATA Device 43
Upgrade URL 43
Resync URL 44
Reboot URL 44
Provisioning Your ATA Device 45
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Provisioning Capabilities 45
Configuration Profile 46
Contents
Chapter 3: Configuring Your System for ITSP Interoperability 47
Network Address Translation (NAT) and Voice over IP (VoIP) 47
NAT Mapping with Session Border Controller 48
NAT Mapping with SIP-ALG Router 48
Configuring NAT Mapping with a Static IP Address 48
Configuring NAT Mapping with STUN 50
Determining the Router’s NAT Mechanism 52
Firewalls and SIP 53
Configuring SIP Timer Values 53
Chapter 4: Configuring Voice Services 54
Supported Codecs 54
Using a FAX Machine 55
Fax Troubleshooting 56
Managing Caller ID Service 58
Silence Suppression and Comfort Noise Generation 60
Configuring Dial Plans 61
About Dial Plans 61
Editing Dial Plans 70
Secure Call Implementation 71
Enabling Secure Calls 71
Secure Call Details 72
Using a Mini-Certificate 73
Generating a Mini Certificate 73
SIP Trunking and Hunt Groups on the SPA8000 75
About SIP Trunking 76
Setting the Trunk Group Call Capacity 78
Inbound Call Routing for a Trunk Group 78
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Contact List for a Trunk Group 79
Outgoing Call Routing for a Trunk Group 81
Configuring a Trunk Group 82
Trunk Group Management 83
Setting the Hunt Policy 85
Additional Notes About Trunk Groups 85
Contents
Chapter 5: Configuring Music on Hold 86
Using the Internal Music Source for Music On Hold 86
Using the Internal Music Source 86
Changing the Music File for the Internal Music Source 87
Configuring a Streaming Audio Server 88
About the Streaming Audio Server 88
Configuring the Streaming Audio Server 90
Using the IVR with an SAS Line 91
Chapter 6: Configuring the PSTN (FXO) Gateway on the SPA3102 92
Connecting to PSTN and VoIP Services 92
How VoIP-To-PSTN Calls Work 93
One-Stage Dialing (SPA3102 and SPA8800) 93
Two-Stage Dialing (SPA3102) 94
How PSTN-To-VoIP Calls Work 95
Terminating Gateway Calls 96
VoIP Outbound Call Routing (SPA3102) 97
Configuring VoIP Failover to PSTN 98
Sharing One VoIP Account Between the FXS and PSTN Lines (SPA3102)98
Other Options 99
PSTN Call to Ring Line 1 (SPA3102) 99
Symmetric RTP (SPA3102 and SPA8800) 99
Call Progress Tones (SPA3102 and SPA8800) 100
Call Scenarios 100
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PSTN to VoIP Call with and Without Ring-Thru 101
VoIP to PSTN Call With and Without Authentication 101
Call Forwarding to PSTN Gateway (SPA3102 and SPA8800) 103
Contents
Appendix A: ATA Routing Field Reference 105
Router Status page 105
Product Information section 106
System Status section 106
WAN Status page 107
Internet Connection Settings section 107
Static IP Settings section 108
PPPoE Settings section 108
Optional Settings section 109
MAC Clone Settings section 110
Remote Management section 110
QOS Settings section 110
VLAN Settings section 111
LAN Status page 111
Networking Service section 111
LAN Networking Settings section 112
Static DHCP Lease Settings section 112
Application page 112
Port Forwarding Settings section 113
DMZ Settings section 113
Miscellaneous Settings section 114
System Reserved Ports Range section 114
Appendix B: ATA Voice Field Reference 115
Info page 116
Product Information section 116
System Status section 117
Line Status section 117
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System Information section (PAP2T) 120
PSTN Line Status section (SPA3102) 120
Trunk Status section (SPA8000) 123
Contents
System page 124
System Configuration section 124
Internet Connection Type section (PAP2T) 125
Optional Network Configuration section (PAP2T) 125
Miscellaneous Settings section (not used with PAP2T) 126
SIP page 127
SIP Parameters section 127
SIP Timer Values (sec) section 129
Response Status Code Handling section 131
RTP Parameters section 132
SDP Payload Types section 134
NAT Support Parameters section 135
Trunking Parameters section (SPA8000) 138
Regional page 139
Call Progress Tones section 140
Distinctive Ring Patterns section 142
Distinctive Call Waiting Tone Patterns section 143
Distinctive Ring/CWT Pattern Names section 144
Ring and Call Waiting Tone Spec section 145
Control Timer Values (sec) section 146
Vertical Service Activation Codes section 148
Vertical Service Announcement Codes section (SPA2102, SPA8000) 154
Outbound Call Codec Selection Codes section 154
Miscellaneous section 156
Line page 160
Line Enable section 161
Streaming Audio Server (SAS) section 162
NAT Settings section 163
Network Settings section 164
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SIP Settings section 165
Call Feature Settings section 168
Proxy and Registration section 169
Subscriber Information section 171
Supplementary Service Subscription section 172
Audio Configuration section 174
Gateway Accounts section (SPA3102) 180
VoIP Fallback to PSTN section (SPA3102 and SPA8800) 180
Dial Plan section 181
FXS Port Polarity Configuration section 183
VoIP-to-PSTN Gateway Setup section (SPA8800) 183
PSTN-To-VoIP Gateway Setup section (SPA8800) 184
FXO Timer Values (sec) section (SPA8800) 185
Contents
PSTN Disconnect Detection section (SPA8800) 186
International Control section (SPA8800) 189
Call Forward, Speed Dial, Supplementary Services, and Ring Settings (SPA8000 and SPA8800) 190
Trunk Group page (SPA8000) 191
Line Enable section 191
Network Settings section 191
SIP Settings section 192
Subscriber Information section 195
Dial Plan section 197
NAT Settings section 197
Proxy and Registration section 198
PSTN Line page (SPA3102) 199
Line Enable section 200
NAT Settings section 200
Network Settings section 201
SIP Settings section 202
Proxy and Registration section 205
Subscriber Information section 206
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Audio Configuration section 207
Dial Plans section 211
VoIP-To-PSTN Gateway Setup section 211
VoIP Users and Passwords (HTTP Authentication) section 213
Ring Settings section 214
FXO (PSTN) Timer Values (sec) section 214
PSTN Disconnect Detection section 216
International Control (Settings) section 219
Contents
User page 221
Call Forward Settings section 222
Selective Call Forward Settings section 223
Speed Dial Settings section 223
Supplementary Service Settings section 224
Distinctive Ring Settings section 225
Ring Settings section 226
PSTN User page (SPA3102) 227
PSTN-To-VoIP Selective Call Forward Settings section 227
PSTN-To-VoIP Speed Dial Settings section 228
PSTN Ring Thru Line 1 Distinctive Ring Settings section 228
PSTN Ring Thru Line 1 Ring Settings section 228
Appendix C: Troubleshooting 229
Appendix D: Environmental Specifications 233
PAP2T 233
SPA2102 234
SPA3102 234
SPA8000 235
SPA8800 235
WRTP54G 236
Appendix E: Where to Go From Here 237
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Contents
Product Resources 237
Related Documentation 238
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About This Document
This guide is intended to help VARs and Service Providers to manage and configure the Cisco Analog Telephone Adapters (ATAs). This preface provides helpful information about this guide and other resources that are available to you. Before you begin to use this guide, refer to the following topics:
• “Purpose,” on page 11
• “Audience,” on page11
• “Document Conventions,” on page12
• “Finding Information in PDF Files,” on page13
Preface
Purpose
Audience
This document provides information that administrators can use to configure and manage Cisco ATAs that are used in conjunction with the SPA9000 Voice System.
This document is written for the following audience:
• Service providers offering services using LVS products
• VARs and resellers who need LVS configuration references
• System administrators or anyone who performs LVS installation and
administration
NOTE This guide does not provide the configuration information required by specific
service providers. Please consult with the service provider for specific service parameters.
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Document Conventions
The following are the typographic conventions used in this document.
Preface
Typographic Element
Boldface
Italic
Monospaced Font
Meaning
May indicate either of the following:
• A user interface element that you need to click, select, or
otherwise act on
• A literal value to be entered in a field.
May indicate either of the following:
• A variable that should be replaced with a literal value.
• The name of a page, section, or field in the user interface
Indicates code samples or system output.
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Finding Information in PDF Files
The SPA9000 Voice System documents are published as PDF files. The PDF Find/ Search tool within Adobe® Reader® lets you find information quickly and easily online. You can perform the following tasks:
• Search an individual PDF file.
• Search multiple PDF files at once (for example, all PDFs in a specific folder or
disk drive).
• Perform advanced searches.
Finding Text in a PDF
Follow this procedure to find text in a PDF file.
STEP 1 Enter your search terms in the Find text box on the toolbar.
Preface
NOTE By default, the Find tool is available at the right end of the Acrobat toolbar. If
the Find tool does not appear, choose Edit > Find.
STEP 2 Optionally, click the arrow next to the Find text box to refine your search by
choosing special options such as Whole Words Only.
STEP 3 Press Enter.
STEP 4 Acrobat displays the first instance of the search term.
STEP 5 Press Enter again to continue to more instances of the term.
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Finding Text in Multiple PDF Files
The
Search
on your PC or local network. The PDF files do not need to be open.
STEP 1 Start Acrobat Professional or Adobe Reader.
window lets you search for terms in multiple PDF files that are stored
Preface
STEP 2 Choose Edit > Search, or click the arrow next to the
Open Full Acrobat Search.
STEP 3 In the
a. Enter the text that you want to find.
b. Choose All PDF Documents in.
c. If you want to specify additional search criteria, click Use Advanced Search
d. Click Search.
Search
From the drop-down box, choose Browse for Location. Then choose the location on your computer or local network, and click OK.
Options, and choose the options you want.
window, complete the following steps:
Find
box and then choose
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Preface
STEP 4 When the Results appear, click + to open a folder, and then click any link to open
the file where the search terms appear.
For more information about the Find and Search functions, see the Adobe Acrobat online help.
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Linksys ATA
Telephone/fax
Ethernet
Broadband CPE
(DSL, cable,
fixed wireless)
Broadband
SIP proxy
Layer 3
IP infrastructure
PSTN
Voice
gateway
187254
V
V
V
Introducing Cisco Small Business Analog Telephone Adapters
This guide describes the administration and use of Cisco Small Business analog telephone adapters (ATAs). These ATA devices are a key element in the end-to­end IP Telephony solution. An ATA device provides user access to Internet phone services through one or more standard telephone RJ-11 phone ports using standard analog telephone equipment. The ATA device connects to a wide area IP network, such as the Internet, through a broadband (DSL or cable) modem or router. The ATA can be used with an onsite call-control system such as the SPA9000 Voice System or legacy PBX, or with an Internet-based call-control system.
1
Figure1 ATA Deployment without Onsite Call Control
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Introducing Cisco Small Business Analog Telephone Adapters
Linksys ATA
Telephone/
fax
Ethernet
Broadband CPE
(DSL, cable,
fixed wireless)
Broadband
SIP proxy
Layer 3
IP infrastructure
PSTN
Voice
gateway
194487
V
V
V
Optional On-Site
Call Control
(SPA9000 or Legacy PBX)
Comparison of ATA Devices
Figure 2 ATA Deployment with Onsite Call Control
This chapter introduces the functionality of the ATA devices and describes the features that are available.
Refer to the following topics:
1
• “Comparison of ATA Devices,” on page 17
• “ATA Connectivity Requirements,” on page 20
• “ATA Software Features,” on page 26
Comparison of ATA Devices
Each ATA device is an intelligent low-density Voice over IP (VoIP) gateway that enables carrier-class residential and business IP Telephony services delivered over broadband or high-speed Internet connections. An ATA device maintains the state of each call it terminates and makes the proper reaction to user input events (such as on/off hook or hook flash). The ATA devices use the Session Initiation Protocol (SIP) open standard so there is little or no involvement by a “middle-man” server or media gateway controller. SIP allows interoperation with all ITSPs that support SIP.
The ATA M ode ls table summarizes the ports and features provided by the ATA devices described in this document.
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Introducing Cisco Small Business Analog Telephone Adapters
Comparison of ATA Devices
ATA M ode ls
1
Product Name
PAP2T 2 — 1 — 2 Voice adapter with two FXS ports.
SPA2102 2 — 1 1 2 Voice adapter with router.
SPA3102 111 1 1Voice adapter with router and PSTN
SPA8000 8 — 1 Maintenance
SPA8800 441Maintenance
WRTP54G2—14 2Wireless-G IP router with two FXS
FXS (Analog Phone)
FXO PSTN
RJ-45 Internet (WAN)
RJ-45 Ethernet (LAN)
only
only
Voice Lines
8 Voice adapter with support for up to
8 Voice adapter with support for up to
Description
connectivity.
eight FXS devices. Supports SIP Trunking for inbound call routing to trunk groups. Also has a single multi­port RJ-21 50-pin connector for optional patch-panel connectivity.
four FXS phones and up to four FXO PSTN lines. Also has a single multi­port RJ-21 50-pin connector for optional patch-panel connectivity.
ports. Provides ATA device functionality.
NOTE The information contained in this guide is not a warranty from Cisco. Customers
planning to use ATA devices in a VoIP service deployment are advised to test all functionality they plan to support before putting the ATA device in service. By implementing ATA devices with the SIP protocol, intelligent endpoints at the edges of a network perform the bulk of the call processing. This allows the deployment of a large network with thousands of subscribers without complicated, expensive servers.
The following figure illustrates how the different ATA devices provide voice connectivity in a VoIP network.
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Introducing Cisco Small Business Analog Telephone Adapters
SPA3102
Broadband router
Broadband
router
SPA8000,
PAP2T
DSL/cable modem
WAG54GP2, AG310
WRP400, RTP300, WRTP54G, and SPA2102
Ethernet/Wireless
LAN
Fax (up to 4
SPA8000)
Analog phone
(up to 8 with
SPA8000)
PSTN
Ethernet/Wired
LAN
Internet
187255
PSTN
Ethernet/Wireless
LAN
Comparison of ATA Devices
Figure 3 How ATAs Provide Voice Connectivity
1
• The PAP2T and the SPA8000 provide FXS ports to connect fax machines and
analog phones to IP telephone services.
• The SPA3102 and the SPA8800 act as SIP-PSTN gateways.
• The WRTP54G router provides ports for analog telephone devices and
provides QoS in the form of priority packet queueing.
NOTE For information about the WRP400, see the WRP400 Administration Guide.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Connectivity Requirements
ATA Connectivity Requirements
An ATA device can be connected to a local router, or directly to the Internet. Each phone connected to an RJ-11 (analog) port on the ATA device connects to other devices through SIP, which is transmitted over the IP network.
In order to ensure connectivity between the devices connected to its FXS ports, the ATA device requires the following functionality to be supplied on the network connected to its Ethernet port:
• Connection to an IP router with hairpinning support
• Connection to an outbound Proxy server
When a phone connected to the ATA device communicates with another phone, it sends a SIP packet onto the internal LAN. The packet is then forwarded to the external LAN or directly to the Internet. The source address and source port on the original packet are assigned by the ATA device DHCP server. The address and port are translated by the ATA device using Network Address Translation (NAT) and Port Address Translation (PAT). The packet is then routed back to the internal network on the ATA device by the local router or the ISP router.
1
Problems can occur with calls between phones connected to the ATA device when an outbound proxy or a router with hairpinning support is not available. The ATA device cannot directly connect the two telephone devices, but requires a local or remote router to route the packet back to its destination on the local network from which it originated.
The necessary routing can be provided by a router with hairpinning support, or by an outbound SIP proxy, which is typically provided by the Internet Telephony Service Provider (ITSP). When relying on the ITSP for interconnecting phones on the ATA device, local phones connected to the ATA device are unable to communicate with each other if the Internet connection is not available for any reason. It is recommended you connect the ATA device to a local router that provides hairpinning support to prevent this problem.
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Introducing Cisco Small Business Analog Telephone Adapters
Line 1 Line 2
Internet
IP Router (with
hairpinning) or
Broadband mode
m
ITSP
ISP
PAP2T
LAN WAN
Ethernet
port
Administrative IVR (Line 1 or
Line 2)
IP
IP
187420
ATA Connectivity Requirements
PAP2T Connectivity
As shown in the following figure, the PAP2T has two FXS ports (voice lines 1 and
2).
1
NOTE
• The IVR functions are accessed by connecting an analog telephone to Line 1.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the PAP2T is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP.
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Introducing Cisco Small Business Analog Telephone Adapters
Administrative I
PC
ATA Connectivity Requirements
SPA2102 Connectivity
As shown in the following illustration, the SPA2102 has two FXS ports (voice lines 1 and 2).
1
NOTE
VR (Line 1 or
Line 2)
SPA
Line 1 Line 2
2102
Ethernet
port
LAN
port
IP Router (with
hairpinning) or
Broadband mode
LAN
m
ISP
WAN
Administration
Internet
IP
IP
ITSP
187257
By default, the device attached to the LAN port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address conflict with a device on the Ethernet port, the network address of the device on the LAN port is automatically changed to 192.168.1.0.
• The IVR functions are accessed by connecting an analog telephone to Line 1.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA2102 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP.
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Introducing Cisco Small Business Analog Telephone Adapters
Line 1 PSTN
Line 1
Internet
IP Router (with
hairpinning) or
Broadband mode
m
ITSP
ISP
SPA
3102
Ethernet
port
LAN
port
LAN WAN
Administrative IVR (Line 1 or
Line 2)
IP
IP
Administration
PC
187259
PSTN
ATA Connectivity Requirements
SPA3102 Connectivity
As shown in the following figure, the SPA3102 has one FXS port (voice line 1).
1
NOTE
By default, the device on the LAN port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address conflict with a device on the Ethernet port, the network address of the device on the LAN port is automatically changed to 192.168.1.0.
• The IVR functions are accessed by connecting an analog telephone to Line 1.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA3102 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP.
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Introducing Cisco Small Business Analog Telephone Adapters
Line 1 Line 2
Internet
IP Router (with
hairpinning) or
Broadband modem
ITSP
ISP
SPA800
0
Line 4
Line 3
Line 6
Line 5
Line 8
Line 7
NAT/PAT
Internal DHCP
server
LAN
WAN
Ethernet
port
AUX
port
Administrative IVR (Line 1 or
Line 2
)
IP
IP
8 FXS (RJ-11/RJ-21 ) ports
Administration
PC
ATA Connectivity Requirements
SPA8000 Connectivity
1
As shown in the following illustration, the SPA8000 consists of eight voice ports (voice lines 1-8).
By default, the device on the AUX port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address conflict with a device on the Ethernet port, the network address of the device on the AUX port is automatically changed to 192.168.1.0.
In the illustration, one fax machine is connected to each pair of ports to illustrate that only one T.38 connection is supported by each of the four pairs of RJ-11 ports.
Up to four fax machines can be connected to the SPA8000 router, but they must be distributed as shown.
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Introducing Cisco Small Business Analog Telephone Adapters
4 FXS (RJ-11/RJ-21) ports
4 FXO ports
PSTN
PSTN
PSTN
PSTN
Line 1
Line 2
Line 3
Line 4
LAN port
Ethernet port
LAN
LAN
IP Router
(with hairpinning)
or Broadband modem
ISP
Internet
ITSP
Administration
PC
Optional On-Site
Call Control
(SPA9000 or Legacy PBX)
PSTN Line 1
PSTN Line 2
PSTN
Line 3
PSTN
Line
235010
ATA Connectivity Requirements
NOTE
• You can use line 1 or line 2 to access the IVR functions.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA8000 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP.
• The SPA8000 is not designed to forward IP packets to devices connected to its
AUX port and that configuration is not supported.
• The SPA8000 also can be configured with trunk groups and trunk lines. See
“SIP Trunking and Hunt Groups on the SPA8000,” on page 75.
1
SPA8800 Connectivity
As shown in the following figure, the SPA8800 has four voice modules that each provide 1 FXS port and 1 FXO port.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
By default, the device on the LAN port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address conflict with a device on the Ethernet port, the network address of the device on the LAN port is automatically changed to 192.168.1.0.
NOTE
• The IVR functions are accessed by connecting an analog telephone to Phone 1
only.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA8800 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP.
1
ATA Software Features
The ATA device is a full featured, fully programmable phone adapter that can be custom provisioned within a wide range of configuration parameters. This section contains a high-level overview of features to provide a basic understanding of the feature breadth and capabilities of the ATA device.
The following sections describe the factors that contribute to voice quality:
• “Voice Supported Codecs,” on page 26
• “SIP Proxy Redundancy,” on page 28
• “Other ATA Software Features,” on page 28
Voice Supported Codecs
Negotiation of the optimal voice codec sometimes depends on the ability of the ATA device to match a codec name with the codec used by the far-end device. The ATA device allows the network administrator to individually name the various codecs that are supported so that the ATA device can successfully negotiate the codec with the far-end equipment. The administrator can select which low-bit-rate codec is to be used for each line. G.711a and G.711u are always enabled.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
You can configure your preferred codec in Configuration Utility. See “SDP
Payload Types section,” on page 134 and “Audio Configuration section,” on page 174. See also “Supported Codecs,” on page 54 for a list of which codecs
are supported on each ATA device.
1
Codec (Voice Compression Algorithm)
G.711 (A-law and mμ-law) This very low complexity codec supports
G.726 This low complexity codec supports compressed 16,
G.729a The ITU G.729 voice coding algorithm is used to
G.723.1 The ATA device supports the use of ITU G.723.1 audio
Description
uncompressed 64 kbps digitized voice transmission at one through ten 5 ms voice frames per packet. This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs.
24, 32, and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides high voice quality.
compress digitized speech. Cisco supports G.729. G.729a is a reduced complexity version of G.729. It requires about half the processing power to code G.729. The G.729 and G.729a bit streams are compatible and interoperable, but not identical.
codec at 6.4 kbps. Up to two channels of G.723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1 simultaneously, or Line 1 or Line 2 can initiate a three-way conference with both call legs using G.723.1.
NOTE When no static payload value is assigned per RFC 1890, the ATA device can
support dynamic payloads for G.726.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
SIP Proxy Redundancy
In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server. An average SIP proxy server may handle thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance. The ATA device supports the use of backup SIP proxy servers (via DNS SRV) so that service disruption should be nearly eliminated.
A relatively simple way to support proxy redundancy is to configure your DNS server with a list of SIP proxy addresses. The ATA device can be instructed to contact a SIP proxy server in a domain named in the SIP message. The ATA device consults the DNS server to get a list of hosts in the given domain that provides SIP services. If an entry exists, the DNS server returns an SRV record that contains a list of SIP proxy servers for the domain, with their host names, priority, listening ports, and so on. The ATA device tries to contact the list of hosts in the order of their stated priority.
1
If the ATA device is currently using a lower priority proxy server, it periodically probes the higher priority proxy to see whether it is back on line, and switches back to the higher priority proxy when possible. SIP Proxy Redundancy is configured in the Line and PSTN Line tabs in the Administration Web Server. See
“ATA Routing Field Reference,” on page 105.
Other ATA Software Features
The following table summarizes other features provided by ATA devices.
Feature Description
Streaming Audio Server
T.38 Fax Relay See “Using a FAX Machine,” on page 55.
Silence Suppression
See “Configuring a Streaming Audio Server,” on
page 88.
See “Silence Suppression and Comfort Noise
Generation,” on page 60.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
Feature Description
•
Modem and Fax Pass-Through
Modem pass-through mode can be triggered only by predialing the number set in the (Set in the Regional tab.)
• FAX pass-through mode is triggered by a CED/CNG tone or
an NSE event.
• Echo canceller is automatically disabled for Modem pass-
through mode.
• Echo canceller is disabled for FAX pass-through if the
parameter for that line (in that case FAX pass-through is the same as Modem pass-through).
• Call waiting and silence suppression is automatically
disabled for both FAX and Modem pass-through. In addition, out-of-band DTMF Tx is disabled during modem or fax pass­through.
FAX Di sable ECA N
1
Modem Line Toggle Code.
(Line 1 or 2 tab) is set to “yes”
Adaptive Jitter Buffer
International Caller ID Delivery
The ATA device can buffer incoming voice packets to minimize out-of-order packet arrival. This process is known as jitter buffering. The jitter buffer size proactively adjusts or adapts in size, depending on changing network conditions.
The ATA device has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the ATA device tries to shrink the jitter buffer over time to achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is lower, it shrinks more quickly.
Adaptive Jitter Buffer is configured in the Line and PSTN Line tabs. See “ATA Voice Field Reference,” on
page 115.
In addition to support of the Bellcore (FSK) and Swedish/ Danish (DTMF) methods of Caller ID (CID) delivery, ATAs provide a large subset of ETSI-compliant methods to support international CID equipment. International CID is configured in the Line and PSTN Line tabs. See “ATA
Voice Field Reference,” on page 115.
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ATA S of t w ar e F ea tu r es
Feature Description
Secure Calls A user (if enabled by service provider or administrator)
has the option to make an outbound call secure in the sense that the audio packets in both directions are encrypted. See “Secure Call Implementation” section
on page 71.
1
Adjustable Audio Frames Per Packet
DTMF The ATA device may relay DTMF digits as out-of-band
Call Progress Tone Generation
This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. See the RTP Packet Size parameter found in the SIP tab in the
“ATA Voice Field Reference,” on page 115.
events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. DTMF is configured in the parameter found in the Line tabs. See the “ATA Voice Field
Reference,” on page 115.
The ATA device has configurable call progress tones. Call progress tones are generated locally on the ATA device so an end user is advised of status (such as ringback). Parameters for each type of tone (for instance a dial tone played back to an end user) may include frequency and amplitude of each component, and cadence information. See the Regional tab in the “ATA Voice Field Reference,”
on page 115.
DTMF Tx Mode
Call Progress Tone Pass Through
Echo Cancellation Impedance mismatch between the telephone and the IP
This feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network. See the Regional tab in the “ATA Voice Field
Reference,” on page 115.
Telephony gateway phone port can lead to near-end echo. The ATA device has a near-end echo canceller that compensates for impedance match. The ATA device also implements an echo suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable. Echo Cancellation is configured in the Regional, Line, and PSTN Line tabs. See “ATA Voice Field
Reference,” on page 115.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
Feature Description
1
Signaling Hook Flash Event
Configurable Dial Plan with Interdigit Time rs
The ATA device can signal hook flash events to the remote party on a connected call. This feature can be used to provide advanced mid-call services with third-party-call­control. Depending on the features that the service provider offers using third-party-call-control, the following ATA features may be disabled to correctly signal a hook­flash event to the softswitch:
•
Call Waiting Service (parameter Line tab)
• Three Way Conference Service (parameter
set in the Line tab)
serv
• Three Way Call Service (parameter
in the Line tab)
You can configure the length of time allowed for detection of a hook flash using the Hook Flash Timer parameter on the Regional tab of the administration web server. See
“ATA Voice Field Reference,” on page 115.
The ATA device has three configurable interdigit timers:
Initial timeout (T)—Signals that the handset is off the hook and that no digit has been pressed yet.
call waiting serv
three-way conf
three-way call serv
set in the
set
Long timeout (L)—Signals the end of a dial string; that is, no more digits are expected.
Short timeout (S)—Used between digits; that is after a digit is pressed a short timeout prevents the digit from being recognized a second time.
See “Configuring Dial Plans,” on page 61 for more information.
Polarity Control The ATA device allows the polarity to be set when a call is
connected and when a call is disconnected. This feature is required to support some pay phone system and answering machines. Polarity Control is configured in the Line and PSTN Line tabs. See “ATA Voice Field
Reference,” on page 115.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
Feature Description
1
Calling Party Control
Report Generation and Event Logging
Syslog and Debug Server Records
Calling Party Control (CPC) signals to the called party equipment that the calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily. This feature is useful for auto­answer equipment, which then knows when to disengage. CPC is configured in the Regional, Line, and PSTN Line tabs. See “ATA Voice Field Reference,” on page 115.
The ATA device reports a variety of status and error reports to assist service providers to diagnose problems and evaluate the performance of their services. The information can be queried by an authorized agent, using HTTP with digested authentication, for instance. The information may be organized as an XML page or HTML page. Report Generation and Event Logging are configured in the System, Line, and PSTN Line tabs. See
“ATA Voice Field Reference,” on page 115.
Syslog and Debug Sever Records log more details than Report Generation and Event Logging. Using the configuration parameters, the ATA device allows you to select which type of activity/events should be logged. Syslog and Debug Server allow the information captured to be sent to a Syslog Server. Syslog and Debug Server Records are configured in the System, Line, and PSTN Line tabs. See “ATA Voice Field Reference,” on
page 115.
SIP Over TCP To guarantee state-oriented communications, you can
choose TCP as the transport protocol for SIP. This protocol is “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. SIP over TCP is configured in the Line tabs. See “ATA Voice Field Reference,” on
page 115.
This feature applies to SPA2102, SPA3102, and SPA8800.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
Feature Description
SIP Over TLS SPA2102, SPA3102, and SPA8800 devices allow the use
of SIP over Transport Layer Security (TLS). SIP over TLS is designed to eliminate the possibility of malicious activity by encrypting the SIP messages of the service provider and the end user. SIP over TLS relies on the widely­deployed and standardized TLS protocol. SIP Over TLS encrypts only the signaling messages and not the media. A separate secure protocol such as Secure Real-Time Transport Protocol (SRTP) can be used to encrypt voice packets. SIP over TLS is configured in the SIP Transport parameter configured in the Line tab(s). See “ATA Vo ice
Field Reference,” on page 115.
Media Loopback SPA2102, SPA3102, and PAP2T devices allow service
providers to use media loopback to quantitatively and qualitatively measure the voice quality experienced by the end user. One device acts as the audio transmitter and receiver while the other device acts as the audio mirror. The audio mirror transmits the audio packets that it receives back to the transmitter/receiver instead of transmitting the data sampled on its local microphone (IP phone) or attached analog telephone (ATA-type device). Media loopback is configured in the User tab. See “ATA
Voice Field Reference,” on page 115.
1
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
Feature Description
1
Register Retry Enhancements
The Register Retry Enhancements feature for SPA2102, SPA3102, and PAP2T devices adds flexibility to the delay timers that are activated when the SIP REGISTER of a device fails. Once a SIP REGISTER failure response code is sent, a delay timer is selected depending on the type of registration failure response code. The delay timers can be one of the following:
•
Reg Retry Random Delay—Random delay range (in seconds) to add to the Register Retry Intvl parameter when retrying a SIP REGISTER after a failure. The default is 0, which disables this feature.
• Reg Retry Long Random Delay—Random delay range (in
seconds) to add to the Register Retry Long Intvl parameter when retrying a SIP REGISTER after a failure. The default is 0, which disables this feature.
• Reg Retry Intvl Cap—The maximum value to cap the
exponential back-off retry delay. The exponential back-off retry delay starts with the setting found in the Register Retry Intvl parameter and doubles it on every REGISTER retry after a failure. In other words, the retry interval after a failure is always set to the seconds configured in the Register Retry Intvl parameter. If this feature is enabled, the Reg Retry Random Delay setting is added on top of the exponential back-off adjusted delay value. The default value is 0, which disables the exponential back-off feature.
Register Retry is configured in the SIP tab. See “ATA Vo ice
Field Reference,” on page 115.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tu r es
Feature Description
1
DHCP Renewal on Time ou t
SPA2102, SPA3102, and PAP2T voice devices typically operate in a network where a DHCP server assigns IP addresses to the devices. Because IP addresses are a limited resource, the DHCP server periodically renews the device lease on the IP address. Therefore, if an ATA device loses its IP address for any reason, or if some other device on the network is assigned its IP address, the communication between the SIP proxy and the device is either severed or degraded.
Whenever an expected SIP response is not received within a programmable amount of time after the corresponding SIP command is sent, the DHCP Renewal on Timeout feature automatically causes the device to request a renewal of its IP address. If the DHCP server returns the IP address that it originally assigned to the device, the ATA device is presumed to be operating correctly. If it returns a different address, the ATA device changes its IP address to the new address provided by the DHCP server. The ATA device then resets, and once again sends a SIP register request for the DHCP server to accept.
Page 36
Basic Administration and Configuration
This chapter describes the equipment and services that are required to install your ATA device and explains how to complete the basic administration and configuration tasks.
Refer to the following topics:
• “Basic Services and Equipment Required” section on page 36
2
• “Downloading Firmware” section on page 37
• “Basic Installation and Configuration” section on page 37
• “Upgrading the Firmware for the ATA Device” section on page 37
• “Setting up Your ATA Device” section on page 38
• “Using the Administration Web Server” section on page 39
• “Upgrading, Rebooting, and Resyncing Your ATA Device” section on
page 43
• “Provisioning Your ATA Device” section on page 45
Basic Services and Equipment Required
To configure your ATA devices, you need the following services and equipment:
• An integrated access device or modem for broadband access to the Internet
• Internet Telephony Service Provider (ITSP) for Voice Over IP Telephone service
• You must have to following information about your account:
• SIP Proxy (IP address or name)
• Account information and Password
Page 37
Basic Administration and Configuration
Downloading Firmware
• Computer with Microsoft Windows XP or Windows Vista (for system
configuration)
• Analog phones
• UPS (uninterruptible Power Source) recommended for devices such as the
Integrated Access Device, network switch, router, and PoE switch to ensure that your phone system continues to work during a power failure, just like your home phone.
Downloading Firmware
Always download and install the latest firmware for your ATA device before doing any configurations. You can find the latest firmware by going to:
2
http://www.cisco.com/en/US/products/ps10024/ tsd_products_support_series_home.html
and clicking the Download Software link.
Basic Installation and Configuration
See your the Quick Installation Guide and the User Guide for the ATA model that you are installing. If you are configuring the complete SPA9000 Voice System, also refer to the documentation for the SPA9000 Voice System.
Upgrading the Firmware for the ATA Device
In this procedure, you install the firmware files that you downloaded previously.
STEP 1 Determine the address of the ATA device:
a. Connect an analog telephone to the Phone 1 or Phone 2 port on the ATA
device. (You may not hear a dial tone. Continue to step b.)
b. Press **** on the keypad to access the IVR menu.
c. Press 110# to determine the Internet (WAN) IP address.
Page 38
Basic Administration and Configuration
Setting up Your ATA Device
STEP 2 Make a note of the IP address that is announced.
NOTE If the administration computer is connected to the Ethernet port of the ATA
device, the default IP address is 192.168.0.1.
STEP 3 Use the administration computer to install the latest firmware:
a. Extract the Zip file, and then run the executable file to upgrade the firmware.
2
b. When the
c. In the next window that appears, enter the IP address of the ATA device, and
then click OK.
d. In the
product number appear. Then click Upgrade.
e. A progress message appears while the upgrade is in progress. The success
window appears when the upgrade is completed. The device reboots.
f. Click OK to close the confirmation message.
g. To verify the upgrade, point the web browser to the IP address of the ATA
device. Check the show the firmware version that you installed.
NOTE You may need to refresh your browser to display the updated page
reflecting the new version number.
Firmware Upgrade Warning
Confirm Upgrade
window, verify that the correct device information and
Router > Status
window appears, click Continue.
page. The
Software Version
field should
Setting up Your ATA Device
After installation and basic configuration of your ATA device, you will use the administration web server to finish your configuration.
ATA devices support two levels of administration privileges: Administrator and User. Both privileges can be password protected.
Page 39
Basic Administration and Configuration
Using the Administration Web Server
NOTE By default, there are no passwords assigned for either the Administrator account or
the User account.
The Administrator account can modify all the web profile parameters and the passwords of both Administrator and User account. The User account can access only part of the web profile parameters. The parameters that the User account can access are specified using the Administrator account on the Provisioning page of the administration web server.
To directly access the Administrator account level privilege, use the following URL:
http://<ipaddress>/admin/voice
If the password has been set for the Administrator account, the browser prompts for authentication. The User account name and the Administrator account name cannot be changed.
2
When browsing pages with the Administrator account privilege, you can switch to User account privilege by clicking the User Login link.
If the User account password is set, the browser prompts for authentication when you click the User Login link. From the User account, you can switch to the Administrator account by clicking the Admin Login link. Authentication is required if the Administrator account password has been set.
NOTE Switching between User and Administrator accounts or between basic and
advanced views discards any uncommitted changes on the web pages.
Using the Administration Web Server
This section describes how to use the administration web server to configure the advanced settings of the ATA device. It includes the following topics:
• “Connecting to the Administration Web Server” section on page 40
• “Setting Up the WAN Configuration for Your ATA Device” section on
page 40
• “Registering to the Service Provider” section on page 42
Page 40
Basic Administration and Configuration
Using the Administration Web Server
• “Advanced Configurations” section on page 43
Connecting to the Administration Web Server
To access the ATA administration web server, perform the following steps.
STEP 1 Start Internet Explorer on a computer that is connected to the same network as the
ATA d evi ce .
STEP 2 Determine the address of the ATA device.
a. Connect an analog telephone to the Phone 1 port of the ATA device. (You may
not hear a dial tone. Continue to step b.)
b. Press **** on the keypad to access the IVR menu.
2
c. Press 110# to determine the Internet (WAN) IP address.
NOTE For more information on the IVR menu, see your Quick Installation Guide or
User Guide for your device, or the LVS Administration Guide.
STEP 3 Direct the browser to the IP address of the ATA device.
STEP 4 The
Router > Status
on to the administrator view by clicking Admin Login, near the top right corner of the page. Then click Advanced.
NOTE By default, no password is required. You can assign an administrative
password later, but it is convenient not to use a password during the initial configuration.
page appears. By default, the page is in Basic User mode. Log
Setting Up the WAN Configuration for Your ATA Device
STEP 1 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
STEP 2 Click Network tab > WAN Setup.
Page 41
Basic Administration and Configuration
Using the Administration Web Server
STEP 3 Complete the WAN configuration for DHCP, static IP addressing, or PPPoE.
For DHCP:
2
a. Select DHCP from the
b. If you use a cable modem, you may need to configure the MAC Clone Settings.
(Contact your ISP for more information.)
c. If your service uses a specific PC MAC address, then select yes from the
Enable MAC Clone Service
d. Then enter the PC’s MAC address in the
For Static IP Addressing:
a. Select Static IP from the
b. In the Static IP Settings section, enter the IP address in the
subnet mask in the
Gateway
c. In the Optional Settings section, enter the DNS server address(es) in the
Primary DNS
For PPPoE:
a. Select PPPoE from the
setting for most DSL users.
field.
and optional
Connection Type
setting.
Connection Type
NetMask
field, and the default gateway IP address in the
Secondary DNS
Connection Type
drop-down menu.
Cloned MAC Address
drop-down menu.
fields.
drop-down menu. This is the correct
field.
Static IP
field, the
b. Enter the values provided by the ITSP in the following fields:
• PPPoE Login Name
• PPPoE Login Password
• PPPoE Service Name
STEP 4 Click Submit All Changes. The ATA device reboots.
STEP 5 To verify your progress, click the Router tab and then click Status. Under
Status Gateway
, confirm the
, and
Primary DNS
WAN Connection Type, Current IP, Current Netmask, Current
.
System
Page 42
Basic Administration and Configuration
Using the Administration Web Server
Registering to the Service Provider
To use VoIP phone service, you must configure your ATA device to the Service Provider.
STEP 1 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
2
STEP 2 Click Voice tab > Line
STEP 3 Enter the account information for your ITSP. The following is the minimum required
configuration to connect the ATA device to an ITSP:
• User ID: The account number or logon name for your ITSP account (Subscriber
Information section)
• Password: The password for your ITSP account (Subscriber Information
section)
• Proxy: The proxy server for your ITSP account (Proxy and Registration section)
STEP 4 After making any necessary changes, click the Submit All Changes button.
STEP 5 To verify your progress, perform the following tasks:
• After the devices reboot, click Voice tab > Info. Scroll down to the
Status
• Use an external phone to place an inbound call to the telephone number that
was assigned by your ITSP. Assuming that you have left the default settings in place, the phone should ring and you can pick up the phone to get two-way audio.
section of the page. Verify that the line is registered.
N
, where N is the line number that you want to configure.
Line 1
• If the line is not registered, you may need to refresh the browser several times
because it can take a few seconds for the registration to succeed. Also verify that your DNS is configured properly.
NOTE If the device has more than one Line tab, each line tab must be configured
separately. Each line tab can be configured for a different ITSP.
Page 43
Basic Administration and Configuration
Upgrading, Rebooting, and Resyncing Your ATA Device
Advanced Configurations
Other parameters may need to be changed from the defaults, depending on the requirements of a specific ITSP. Some of the commonly configured parameters include the following:
• Streaming Audio Server—You can enable an external music source for music
on hold. See the “Configuring a Streaming Audio Server,” on page 88 for further information.
• NAT Settings—You can adjust these settings to resolve issues that arise when
using a ATA on a network behind a Network Address Translation (NAT) device. See the “Network Address Translation (NAT) and Voice over IP (VoIP),” on
page 47 for further information.
• Subscriber Information—You can configure security parameters. See the
“Secure Call Implementation,” on page 71 for further information.
2
• Dial Plan—You can configure a dial plan for a specific line. See the
“Configuring Dial Plans,” on page 61 for further information.
Upgrading, Rebooting, and Resyncing Your ATA Device
The administration web server supports upgrading, rebooting, and resyncing functions through special URLs. Administrator account privilege is needed for these functions.
Upgrade URL
The Upgrade URL lets you upgrade the ATA device to the firmware specified by the URL, which can identify either a TFTP or HTTP server.
NOTE If the value of the
cannot upgrade the ATA device even if the web page indicates otherwise.
Upgrade Enable
parameter in the Provisioning page is No, you
The syntax of the Upgrade URL is as follows:
http://spa-ip-addr/admin/upgrade?[protocol://][server-name[:port]][/ firmware-pathname]
Both HTTP and TFTP are supported for the upgrade operation.
Page 44
Basic Administration and Configuration
Upgrading, Rebooting, and Resyncing Your ATA Device
If no
protocol
host that requests the URL is used as
If no port specified, the default port of the protocol is used. (69 for TFTP or 80 for HTTP)
The
firmware-pathname
directory on the TFTP or HTTP server. If no
spa.bin
http://192.168.2.217/admin/upgrade?tftp://192.168.2.251/spa.bin
is specified, TFTP is assumed. If no
is assumed, as in the following example:
Resync URL
The Resync URL lets you force the ATA device to do a resync to a profile specified in the URL, which can identify either a TFTP, HTTP, or HTTPS server. The syntax of the Resync URL is as follows:
server-name
server-name
is typically the file name of the binary located in a
.
firmware-pathname
is specified, the
is specified,
2
/
http://spa-ip-addr/admin/resync?[[protocol://][server-name[:port]]/profile­pathname]
NOTE The SPA resyncs only when it is idle.
If no parameter follows page is used.
protocol
If no host that requests the URL is used as
If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, and 443 for HTTPS).
The profile-path is the path to the new profile with which to resync, for example:
http://192.168.2.217admin/resync?tftp://192.168.2.251/spaconf.cfg
is specified, TFTP is assumed. If no
/resync?,
the Profile Rule setting from the Provisioning
Reboot URL
The Reboot URL lets you reboot the ATA device. The Reboot URL is as follows:
server-name
server-name
.
is specified, the
http://spa-ip-addr/admin/reboot
NOTE The ATA device reboots only when it is idle.
Page 45
Basic Administration and Configuration
Provisioning Your ATA Device
Provisioning Your ATA Device
This section describes the provisioning functionality of the ATA device. This section includes the following topics:
• “Provisioning Capabilities” section on page 45
• “Configuration Profile” section on page 46
For detailed information about provisioning your ATA device, refer to the SPA
Provisioning Guide.
Provisioning Capabilities
The ATA device provides for secure provisioning and remote upgrade. Provisioning is achieved through configuration profiles transferred to the device via TFTP, HTTP, or HTTPS. To configure Provisioning, go to Provisioning tab in the administration web server.
2
The ATA device can be configured to automatically resync its internal configuration state to a remote profile periodically and on power up. The automatic resyncs are controlled by configuring the desired profile URL into the device.
The ATA device accepts profiles in XML format, or alternatively in a proprietary binary format, which is generated by a profile compiler tool available from Cisco. Find the Profiler Compiler for your ATA at http://www.cisco.com/web/partners/
sell/smb/products/voice_and_conferencing.html#~vc_technical_resources.
The ATA device supports up to 256-bit symmetric key encryption of profiles. For the initial transfer of the profile encryption key (initial provisioning stage), the ATA device can receive a profile from an encrypted channel (HTTPS), or it can resync to a binary profile generated by the Cisco-supplied profile compiler. In the latter case, the profile compiler can encrypt the profile specifically for the target ATA device, without requiring an explicit key exchange.
Remote firmware upgrade is achieved via TFTP or HTTP (firmware upgrades using HTTPS are not supported). Remote upgrades are controlled by configuring the desired firmware image URL into the ATA device via a remote profile resync.
For further information about remote provisioning refer to the SPA Provisioning Guide.
Page 46
Basic Administration and Configuration
Provisioning Your ATA Device
Configuration Profile
The ATA configuration profile can be either an XML file or a binary file with a proprietary format.
The XML file consists of a series of elements (one per configuration parameter), encapsulated within the element tags <flat-profile> … </flat-profile>. The encapsulated elements specify values for individual parameters. Here is an example of a valid XML profile:
<flat-profile> <Admin_Passwd>some secret</Admin_Passwd> <Upgrade_Enable>Yes</Upgrade_Enable> </flat-profile>
Binary format profiles contain ATA parameter values and user access permissions for the parameters. By convention, the profile uses the extension .cfg (for example, spa2102.cfg). The Profile Compiler (SPC) tool compiles a plain-text file containing parameter-value pairs into a properly formatted and encrypted .cfg file. The SPC tool is available for the Win32 environment and Linux-i386-elf environment. Requests for SPC tools compiled on other platforms are evaluated on a case-by­case basis. Please contact your sales representative for further information about obtaining the SPC tool.
2
The syntax of the plain-text file accepted by the profile compiler is a series of parameter-value pairs, with the value in double quotes. Each parameter-value pair is followed by a semicolon. Here is an example of a valid text source profile for input to the SPC tool:
Admin_Passwd “some secret”; Upgrade_Enable “Yes”;
Refer to the SPA Provisioning Guide for further details.
The names of parameters in XML profiles can generally be inferred from the ATA configuration Web pages, by substituting underscores (_) for spaces and other control characters. Further, to distinguish between Lines 1, 2, 3, and 4, corresponding parameter names are augmented by the strings _1_, _2_, _3_, and _4_. For example, Line 1 Proxy is named Proxy_1_ in XML profiles.
Parameters in the case of source text files for the SPC tool are similarly named, except that to differentiate Line 1, 2, 3, and 4, the appended strings ([1], [2], [3], or [4]) are used. For example, the Line 1 Proxy is named Proxy[1] in source text profiles for input to the SPC.
Page 47
Configuring Your System for ITSP Interoperability
This chapter provides configuration details to help you to ensure that your infrastructure properly supports voice services.
• “Network Address Translation (NAT) and Voice over IP (VoIP),” on page 47
• “Firewalls and SIP,” on page 53
3
• “Configuring SIP Timer Values,” on page 53
Network Address Translation (NAT) and Voice over IP (VoIP)
NAT is a function that allows multiple devices to share the same public, routable, IP address to establish connections over the Internet. NAT is present in many broadband access devices to translate public and private IP addresses. To enable VoIP to co-exist with NAT, some form of NAT traversal is required.
Some ITSPs provide NAT traversal, but some do not. If your ITSP does not provide NAT traversal, you have several options.
• “NAT Mapping with Session Border Controller,” on page 48
• “NAT Mapping with SIP-ALG Router,” on page 48
• “Configuring NAT Mapping with a Static IP Address,” on page 48
• “Configuring NAT Mapping with STUN,” on page 50
Page 48
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
NAT Mapping with Session Border Controller
It is strongly recommended that you choose an ITSP that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the ITSP, you have more choices in selecting a router.
NAT Mapping with SIP-ALG Router
If the ITSP network does not provide a Session Border Controller functionality, you can achieve NAT mapping by using a router that has a SIP ALG (Application Layer Gateway). The WRV200 router is recommended for this purpose, although any router with a SIP-ALG can be used. By using a SIP-ALG router, you have more choices in selecting an ITSP.
3
Configuring NAT Mapping with a Static IP Address
If the ITSP network does not provide a Session Border Controller functionality, and if other requirements are met, you can configure NAT mapping to ensure interoperability with the ITSP.
Requirements:
• You must have an external (public) IP address that is static.
• The NAT mechanism used in the router must be symmetric. See “Determining
the Router’s NAT Mechanism,” on page 52.
• The LAN switch must be configured to enable Spanning Tree Protocol and Port
Fast on the ports to which the SPA devices are connected.
NOTE Use NAT mapping only if the ITSP network does not provide a Session Border
Controller functionality.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > SIP.
Page 49
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
STEP 3 Scroll down to the NAT Support Parameters section, and then enter the following
settings to support static mapping to your public IP address:
• Handle VIA received, Insert VIA received, Substitute VIA Addr: yes
• Handle VIA rport, Insert VIA rport, Send Resp To Src Port: yes
• EXT IP: Enter the public IP address for your router.
Voice tab > SIP: NAT Support Parameters
3
STEP 4
STEP 5 Scroll down to the NAT Settings section.
Click Voice tab > Line N, where N represents the line interface number.
• NAT Mapping Enable: Choose YES.
• NAT Keep Alive Enable: Choose YES (optional).
Voice tab > Line N > NAT Settings
STEP 6
Click Submit All Changes.
NOTE You also need to configure the firewall settings on your router to allow SIP
traffic. See “Firewalls and SIP,” on page 53.
Page 50
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Configuring NAT Mapping with STUN
If the ITSP network does not provide a Session Border Controller functionality, and if other requirements are met, it is possible to use STUN as a mechanism to discover the NAT mapping. This option is considered a practice of last resort and should be used only if the other methods are unavailable.
Requirements:
• STUN is a viable option only if your router uses asymmetric NAT. See
“Determining the Router’s NAT Mechanism,” on page 52.
• You must have a computer running STUN server software.
• The LAN switch must be configured to enable Spanning Tree Protocol and Port
Fast on the ports to which the SPA devices are connected.
3
NOTE Use NAT mapping only if the ITSP network does not provide a Session Border
Controller functionality.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > SIP.
STEP 3 Scroll down to the NAT Support Parameters section, and then enter the following
settings to enable and support the STUN server settings:
• Handle VIA received: yes
• Handle VIA rport: yes
• Insert VIA received: yes
• Insert VIA rport: yes
• Substitute VIA Addr: yes
• Send Resp To Src Port: yes
• STUN Enable: Choose yes.
• STUN Server: Enter the IP address for your STUN server.
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Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Voice tab > SIP > NAT Support Parameters
3
STEP 4
STEP 5 Scroll down to the NAT Settings section.
Click Voice tab > Line N, where N is the number of the line interface.
• NAT Mapping Enable: Choose yes.
• NAT Keep Alive Enable: Choose yes (optional).
Voice tab > Line N > NAT Settings
NOTE Your ITSP may require the SPA device to send NAT keep alive messages to
keep the NAT ports open permanently. Check with your ITSP to determine the requirements.
STEP 6 Click Submit All Changes.
NOTE You also need to configure the firewall settings on your router to allow SIP
traffic. See “Firewalls and SIP,” on page 53.
Page 52
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Determining the Router’s NAT Mechanism
STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from one internal IP address and port to one external, routable destination IP address and port. If another packet is sent from the same source IP address and port to a different destination, then a different IP address and port number combination is used. This method is restrictive because an external host can send a packet to a particular port on the internal host only if the internal host first sent a packet from that port to the external host.
NOTE This procedure assumes that a syslog server is configured and is ready to receive
syslog messages.
STEP 1 Make sure you do not have firewall running on your PC that could block the syslog
port (port 514 by default).
3
STEP 2 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 3 To enable debugging, complete the following tasks:
a. Click Voice tab > System.
b. In the Debug Server field, enter the IP address of your syslog server. This
address and port number must be reachable from the SPA9000.
c. From the Debug level drop-down list, choose 3.
STEP 4 To collect information about the type of NAT your router is using, complete the
following tasks:
a. Click Voice tab > SIP.
b. Scroll down to the NAT Support Parameters section.
c. From the STUN Test Enable field, choose yes.
STEP 5 To enable SIP signalling, complete the following task:
a. Click Voice tab > Line
N
, where N represents the line interface number.
b. In the SIP Settings section, choose full from the SIP Debug Option field.
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Configuring Your System for ITSP Interoperability
Firewalls and SIP
STEP 6 Click Submit All Changes.
STEP 7 View the syslog messages to determine whether your network uses symmetric
NAT. Look for a warning header in the REGISTER messages, such as Warning: 399 spa "Full Cone NAT Detected.”
Firewalls and SIP
To enable SIP requests and responses to be exchanged with the SIP proxy at the ITSP, you must ensure that your firewall allows both SIP and RTP unimpeded access to the Internet.
• Make sure that the following ports are not blocked:
3
• SIP ports—UDP port 5060 through 5063, which are used for the ITSP line
interfaces
• RTP ports—16384 to 16482
• Also disable SPI (Stateful Packet Inspection) if this function exists on your
firewall.
Configuring SIP Timer Values
The default timer values should be adequate in most circumstances. However, you can adjust the SIP timer values as needed to ensure interoperability with your ISTP. For example, if SIP requests are returned with an “invalid certificate” message, you may need to enter a longer SIP T1 retry value.
To view the default settings or to make changes, open the Voice > SIP page, and scroll down to the SIP Timer Values section. For field descriptions, see ”SIP
Timer Values (sec) section,” on page 129 of Appendix B.
Page 54
Configuring Voice Services
This chapter describes how to configure your ATA device to meet the customer’s requirements for voice services.
• “Supported Codecs,” on page 54
• “Using a FAX Machine,” on page 55
• “Managing Caller ID Service,” on page 58
4
• “Silence Suppression and Comfort Noise Generation,” on page 60
• “Configuring Dial Plans,” on page 61
• “Secure Call Implementation,” on page 71
• “SIP Trunking and Hunt Groups on the SPA8000,” on page 75
Supported Codecs
The following list shows the current supported codecs for each ATA device. If you need to change the G711u codec which is configured by default, set your preferred codecs in the FXS Line tab(s); Audio Configuration. You may set your first, second, and third preferred codec. See “ATA Routing Field Reference,” on
page 105.
PAP2T / SPA2102 / SPA3102 / SPA8000 / SPA8800
• G.711u (configured by default)
• G.711a
• G.726-16
• G.726-24
• G.726-32
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Configuring Voice Services
Using a FAX Machine
• G.726-40
• G.729a
• G.723
WRTP54G
• G.711u (configured by default)
• G.711a
• G.726-32
• G.729a
• G.723
4
Using a FAX Machine
You can connect a fax machine to an FXS port on the SPA2102, SPA3102, SPA8000, and SPA8800. T.38 Fax is supported on these devices, for fax transmission over an IP network. The SPA2102 and SPA3102 support a single connection, while the SPA8000 and SPA8800 support one connection for each pair of FXS ports (1/2, 3/4, 5/6, and 7/8) for a maximum of four connections.
Follow this procedure to optimize fax completion rates.
STEP 1 Upgrade the ATA firmware to the latest version
STEP 2 Ensure that you have enough bandwidth for uplink and downlink.
• For G.711 fallback, it is recommend to have approximately 100Kbps.
• For T.38, allocate at least 50 kbps.
STEP 3 To optimize G.711 fallback fax completion rates, set the following on the Line tab
of your ATA device:
• Network Jitter Buffer: very high
• Jitter buffer adjustment: disable
• Call Waiting: no
• 3 Way Calling: no
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Configuring Voice Services
Using a FAX Machine
• Echo Canceller: no
• Silence suppression: no
• Preferred Codec: G.711
• Use pref. codec only: yes
STEP 4 If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax
relay) and enable fax using modem passthrough.
For example:
modem passthrough nse payload-type 110 codec g711ulaw fax rate disable fax protocol pass-through g711ulaw
STEP 5 Enable T.38 fax on the SPA 2102 by configuring the following parameter on the
Line tab for the FXS port to which the FAX machine is connected:
4
FAX_Passthru_Method: ReINVITE
NOTE If a T.38 call cannot be set-up, then the call should automatically revert to
G.711 fallback.
STEP 6 If you are using a Cisco media gateway use the following settings:
Make sure the Cisco gateway is correctly configured for T.38 with the SPA dial peer. For example:
fax protocol T38 fax rate voice fax-relay ecm disable fax nsf 000000 no vad
Fax Troubleshooting
If have problems sending or receiving faxes, complete the following steps:
STEP 1 Verify that your fax machine is set to a speed between 7200 and 14400.
STEP 2 Send a test fax in a controlled environment between two ATAs.
STEP 3 Determine the success rate.
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Configuring Voice Services
Using a FAX Machine
STEP 4 Monitor the network and record the following statistics:
• Jitter
• Loss
• Delay
STEP 5 If faxes fail consistently, capture a copy of the web interface settings by selecting
Save As > Web page, complete from the administration web server page. You
can send this configuration file to Technical Support.
STEP 6 Enable and capture the debug log. For instructions, refer to Appendix C,
“Troubleshooting.”.
NOTE You may also capture data using a sniffer trace.
4
STEP 7 Identify the type of fax machine connected to the ATA device.
STEP 8 Contact technical support:
• If you are an end user of VoIP products, contact the reseller or Internet
telephony service provider (ITSP) that supplied the equipment.
• If you are an authorized Cisco partner, contact Cisco technical support.
Page 58
Configuring Voice Services
Managing Caller ID Service
Managing Caller ID Service
The choice of caller ID (CID) method is dependent on your area/region. To configure CID, use the following parameters:
Parameter Ta b Description and Value
4
Caller ID Method
Regional The following choices are available:
•
Bellcore (N.Amer,China)—CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).
• DTMF (Finland, Sweden)—CID only. DTMF sent after
polarity reversal (and no DTAS) and before first ring.
• DTMF (Denmark)—CID only. DTMF sentbefore first
ring with no polarity reversal and no DTAS.
• ETSI DTMF—CID only. DTMF sent after DTAS (and no
polarity reversal) and before first ring.
• ETSI DTMF With PR—CID only. DTMF sent after
polarity reversal and DTAS and before first ring.
• ETSI DTMF After Ring—CID only. DTMF sent after
first ring (no polarity reversal or DTAS).
• ETSI FSK—CID, CIDCW, and VMWI. FSK sent after
DTAS (but no polarity reversal) and before first ring. Waits for ACK from CPE after DTAS for CIDCW.
• ETSI FSK With PR (UK)—CID, CIDCW, and VMWI.
FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from CPE after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.
Caller ID FSK Standard
Regional
• DTMF (Denmark) With PR—CID only. DTMF sent after
polarity reversal (and no DTAS) and before first ring.
The default is Bellcore(N.Amer, China).
The ATA device supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.2 3.
The default is bell 202.
This field is not found in the PAP2T.
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Configuring Voice Services
Polarity
Reversal
First
Ring
CAS
(DTAS)
DTMF/
FSK
Polarity
Reversal
CAS
(DTAS)
FSK
CAS
(DTAS)
Wait For
ACK
FSK
First Ring
FSK
OSI FSK
a) Bellcore/ETSI Onhook Post-Ring FSK
d) Bellcore Onhook FSK w/o Ring
f) Bellcore/ETSI Offhook FSK
c) ETSI Onhook Pre-Ring FSK/DTMF
e) ETSI Onhook FSK w/o Ring
DTMF
b) ETSI Onhook Post-Ring DTMF
First Ring
Managing Caller ID Service
There are three types of Caller ID:
4
• On Hook Caller ID Associated with Ringing — This type of Caller ID is used
for incoming calls when the attached phone is on hook. See the following figure (a) – (c). All CID methods can be applied for this type of CID.
• On Hook Caller ID Not Associated with Ringing — This feature is used to
send VMWI signal to the phone to turn the message waiting light on and off (see Figure 1 (d) and (e)). This is available only for FSK-based CID methods: (Bellcore, ETSI FSK, and ETSI FSK With PR).
• Off Hook Caller ID — This is used to delivery caller-id on incoming calls
when the attached phone is off hook (see the following figure). This can be call waiting caller ID (CIDCW) or to notify the user that the far end party identity has changed or updated (such as due to a call transfer). This is available only for FSK-based CID methods: (Bellcore, ETSI FSK, and ETSI FSK With PR).
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Configuring Voice Services
Silence Suppression and Comfort Noise Generation
Silence Suppression and Comfort Noise Generation
Voice Activity Detection (VAD) with Silence Suppression is a means of increasing the number of calls supported by the network by reducing the required bandwidth for a single call. VAD uses a sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether or not speech is present. If the VAD algorithm decides speech is not present, the silence suppression and comfort noise generation is activated. This is accomplished by removing and not transmitting the natural silence that occurs in normal two-way connection. The IP bandwidth is used only when someone is speaking. During the silent periods of a telephone call, additional bandwidth is available for other voice calls or data traffic because the silence packets are not being transmitted across the network.
Comfort Noise Generation provides artificially-generated background white noise (sounds), designed to reassure callers that their calls are still connected during silent periods. If Comfort Noise Generation is not used, the caller may think the call has been disconnected because of the “dead silence” periods created by the VAD and Silence Suppression feature.
4
Silence suppression is configured in the Line and PSTN Line tabs. See “ATA
Routing Field Reference,” on page 105.
Page 61
Configuring Voice Services
Configuring Dial Plans
Configuring Dial Plans
Dial plans determine how the digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls such as long distance or international.
This section includes information that you need to understand dial plans, as well as procedures for configuring your own dial plans. This section includes the following topics:
• “About Dial Plans,” on page 61
• “Editing Dial Plans,” on page 70
4
About Dial Plans
This section provides information to help you understand how dial plans are implemented.
Refer to the following topics:
• “Digit Sequences,” on page 61
• “Digit Sequence Examples,” on page 64
• “Acceptance and Transmission the Dialed Digits,” on page 66
• “Dial Plan Timer (Off-Hook Timer),” on page 67
• “Interdigit Long Timer (Incomplete Entry Timer),” on page 68
• “Interdigit Short Timer (Complete Entry Timer),” on page 68
Digit Sequences
A dial plan contains a series of digit sequences, separated by the | character. The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan consists of a series of elements, which are individually matched to the keys that the user presses.
NOTE White space is ignored, but may be used for readability.
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Configuring Voice Services
Configuring Dial Plans
Digit Sequence Function
4
0 1 2 3 4 5 6 7 8 9 0 * #
x Enter x to represent any character on the phone
[sequence] Enter characters within square brackets to create
.
(period)
<dialed:substituted> For sequence substitution, use this format to
Enter any of these characters to represent a key that the user must press on the phone keypad.
keypad.
a list of accepted key presses. The user can press any one of the keys in the list.
Numeric range
•
For example, you would enter user to press any one digit from 2 through 9.
• Numeric range with other characters
For example, you would enter the user to press 3, 5, 6, 7, 8, or *.
Enter a period for element repetition. The dial plan accepts 0 or more entries of the digit. For example, 01. allows users to enter 0, 01, 011, 0111, and so on.
indicate that certain dialed digits are replaced by other characters when the sequence is transmitted. The dialed digits can be zero or more characters.
[2-9] to allow the
[35-8*] to allow
EXAMPLE 1: <8:1650>xxxxxxx
When the user presses 8 followed by a seven­digit number, the system automatically replaces the dialed 8 with 1650. If the user dials
85550112, the system transmits 16505550112. EXAMPLE 2: <:1>xxxxxxxxxx
In this example, no digits are replaced. When the user enters a 10-digit string of numbers, the number 1 is added at the beginning of the sequence. If the user dials 9725550112, the system transmits 19725550112
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Configuring Voice Services
Configuring Dial Plans
Digit Sequence Function
4
,
(comma)
!
(exclamation point)
*xx
S0 or L0
For an intersequence tone, enter a comma between digits to play an “outside line” dial tone after a user-entered sequence.
EXAMPLE: 9, 1xxxxxxxxxx
An “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1.
For number barring, enter an exclamation point to prohibit a dial sequence pattern.
EXAMPLE: 1900xxxxxxx!
The system rejects any 11-digit sequence that begins with 1900.
Enter an asterisk to allow the user to enter a 2­digit star code.
For Interdigit Timer Master Override, enter S0 to reduce the short inter-digit timer to 0 seconds, or enter L0 to reduce the long inter-digit timer to 0 seconds.
P
NOTE The SPA implicitly appends the vertical code sequences entered in the regional
parameter settings to the end of the dial plan. Likewise, if Enable_IP_Dialing is enabled, then IP dialing is also accepted on the associated line.
For a pause, enter P followed by a number and a space. The duration of the pause is the specified number of seconds. This feature is typically used for implementation of a hot line and warm line, with 0 delay for the hot line and a non-zero delay for a warm line.
EXAMPLE: P5
REVIEW COMMENT: need example
Page 64
Configuring Voice Services
Configuring Dial Plans
Digit Sequence Examples
The following examples show digit sequences that you can enter in a dial plan.
In a complete dial plan entry, sequences are separated by a pipe character (|), and the entire set of sequences is enclosed within parentheses.
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxx xxx | 9, 1 900 xxxxxx x ! | 9, 011xxxxxx. | 0 | [49]11 )
• Extensions on your system
4
EXAMPLE: (
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
[1-8]xx Allows a user dial any three-digit number that starts with the digits 1
through 8. If your system uses four-digit extensions, you would instead enter the following string:
[1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
[1-8]xx x
• Local dialing with seven-digit number
EXAMPLE: ( [1-8]xx | <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)
9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can
enter any seven-digit number, as in a local call.
9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
• Local dialing with 3-digit area code and a 7-digit local number
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required.
After a user presses 9, an external dial tone sounds. The user must enter a 10­digit number that begins with a digit 2 through 9. The system automatically inserts the 1 prefix before transmitting the number to the carrier.
9, <:1>[2-9]xxxxxxxxx | 8,
• Local dialing with an automatically inserted 3-digit area code
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2- 9]xxxxxxxxx |
<:1212>xxxxxxx
011xxxxxx. | 0 | [49]11 )
8,
| 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,
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Configuring Voice Services
Configuring Dial Plans
• U.S. long distance dialing
• Blocked number
4
8, <:1212>xxxxxxx This is example is useful where a local area code is required
by the carrier but the majority of calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number. The system automatically inserts the 1 prefix and the 212 area code before transmitting the number to the carrier.
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 011xxxxxx. | 0 | [49]11 )
9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The
user can enter any 11-digit number that starts with 1 and is followed by a digit 2 through 9.
9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxx xx ! | 9,
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 011xxxxxx. | 0 | [49]11 )
9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from
dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S.. After the user press 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected.
9, 1 900 xxxxxxx ! |
• U.S. international dialing
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [ 2-9] xxxxxxxxx | 9, 1 900 xx xxxxx ! |
9, 011xxxxxx. | 0 | [49]11 )
9, 01 1xxxxxx. After the user presses 9, an external dial tone sounds. The user can
enter any number that starts with 011, as in an international call from the U.S.
• Informational numbers
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. |
0 | [49]11 )
9, <:1>[2-9]xxxx xxxxx | 8,
0 | [49]11 This example includes two digit sequences, separated by the pipe
character. The first sequence allows a user to dial 0 for an operator. The second sequence allows the user to enter 411 for local information or 911 for emergency services.
Page 66
Configuring Voice Services
Configuring Dial Plans
Acceptance and Transmission the Dialed Digits
When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As more digits are entered by the user, the set of candidates diminishes until only one or none are valid. When a terminating event occurs, the SPA9000 either accepts the user-dialed sequence and initiates a call, or else rejects the sequence as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is invalid.
The following table explains how terminating events are processed.
Ter min at ing Eve nt Processing
4
The dialed digits do not match any sequence in the dial plan.
The dialed digits exactly match one sequence in the dial plan.
A timeout occurs. The number is rejected if the dialed digits are
The user presses the # key or the dial softkey on the phone display.
The number is rejected.
•
If the sequence is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan.
• If the sequence is blocked by the dial plan, the
number is rejected.
not matched to a digit sequence in the dial plan within the time specified by the applicable interdigit timer.
The Interdigit Long Timer applies when the
•
dialed digits do not match any digit sequence in the dial plan. The default value is 10 seconds.
• The Interdigit Short Timer applies when the
dialed digits match one or more candidate sequences in the dial plan. The default value is 3 seconds.
If the sequence is complete and is allowed by
•
the dial plan, the number is accepted and is transmitted according to the dial plan.
• If the sequence is incomplete or is blocked by
the dial plan, the number is rejected.
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Configuring Voice Services
Configuring Dial Plans
Dial Plan Timer (Off-Hook Timer)
You can think of the Dial Plan Timer as “the off-hook timer.” This timer starts counting when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated. Unless you have a special dial plan string to allow a null entry, the call is rejected. The default value is 5.
Syntax for the Dial Plan Timer
SYNTAX: (P s<:n> | dial plan )
• s: The number of seconds; if no number is entered after P, the default timer of 5
• n: (optional): The number to transmit automatically when the timer expires; you
4
seconds applies.
can enter an extension number or a DID number. No wildcard characters are allowed because the number will be transmitted as shown. If you omit the number substitution, <:n>, then the user hears a reorder (fast busy) tone after the specified number of seconds.
Examples for the Dial Plan Timer
• Allow more time for users to start dialing after taking a phone off hook.
EXAMPLE: ( | 9,8,011xx. | 9,8,xx.|[1-8]xx )
P9 After taking a phone off hook, a user has 9 seconds to begin dialing. If no
digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone. By setting a longer timer, you allow more time for users to enter the digits.
P9 | (9,8<:1408>[2-9]xxxxx x | 9,8,1[2-9]xxxxxxxxx
• Create a hotline for all sequences on the System Dial Plan
EXAMPLE: ( 9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
P9<:23> After taking the phone off hook, a user has 9 seconds to begin dialing. If
no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23.
P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-
• Create a hotline on a line button for an extension
EXAMPLE:
( P0 <:1000>)
With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook. Enter this sequence in the Phone Dial Plan for Ext 2 or higher on a client station.
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Configuring Voice Services
Configuring Dial Plans
Interdigit Long Timer (Incomplete Entry Timer)
You can think of this timer as the “incomplete entry” timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. The default value is 10 seconds.
NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can
modify the Control Timer that controls the default interdigit timers for all calls. See
“Resetting the Control Timers,” on page 70.
Syntax for the Interdigit Long Timer
SYNTAX: L:s, ( dial plan )
4
• s: The number of seconds; if no number is entered after L:, the default timer of
5 seconds applies.
• Note that the timer sequence appears to the left of the initial parenthesis for the
dial plan.
Example for the Interdigit Long Timer
EXAMPLE: L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx |
9,8,011xx. | 9,8,xx.|[1-8 ]xx)
L:15, This dial plan allows the user to pause for up to 15 seconds between digits
before the Interdigit Long Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.
Interdigit Short Timer (Complete Entry Timer)
You can think of this timer as the “complete entry” timer. This timer measures the interval between dialed digits. It applies when the dialed digits match at least one digit sequence in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated. If it is valid, the call proceeds. If it is invalid, the call is rejected. The default value is 3 seconds.
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Configuring Voice Services
Configuring Dial Plans
Syntax for the Interdigit Short Timer
• SYNTAX 1: S:s, ( dial p lan )
• SYNTAX 2: sequence Ss
Examples for the Interdigit Short Timer
• Set the timer for the entire dial plan.
4
Use this syntax to apply the new setting to the entire dial plan within the parentheses.
Use this syntax to apply the new setting to a particular dialing sequence.
s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies.
EXAMPLE:
9,8,011xx. | 9,8,xx.|[1-8 ]xx)
S:6, While entering a number with the phone off hook, a user can pause for up
to 15 seconds between digits before the Interdigit Short Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.
S:6, (9,8<:1408>[2-9]xxxx xx | 9,8,1[2-9]xxxxxxxxx |
• Set an instant timer for a particular sequence within the dial plan.
EXAMPLE: (9,8<:1408>[2-9]xxxxxx | 9,8,011xx. | 9,8,xx.|[1-8 ]xx)
9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted automatically
when the user dials the final digit in the sequence.
9,8,1[2-9]xxxxxxxxxS0 |
Page 70
Configuring Voice Services
Configuring Dial Plans
Editing Dial Plans
You can edit dial plans and can modify the control timers.
STEP 1 Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.
Entering the Line Interface Dial Plan
This dial plan is used to strip steering digits from a dialed number before it is transmitted out to the carrier.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
4
STEP 2 Click Voice tab > Line
STEP 3 Scroll down to the Dial Plan section.
STEP 4 Enter the digit sequences in the Dial Plan field. For more information, see “A bo ut
Dial Plans,” on page 61.
STEP 5 Click Submit All Changes.
N
, where N represents the line interface number.
Resetting the Control Timers
You can use the following procedure to reset the default timer settings for all calls.
NOTE If you need to edit a timer setting only for a particular digit sequence or type of call,
you can edit the dial plan. See “About Dial Plans,” on page 61.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > Regional.
STEP 3 Scroll down to the Control Timer Values section.
STEP 4 Enter the desired values in the Interdigit Long Timer field and the Interdigit Short
Timer field. Refer to the definitions at the beginning of this section.
Page 71
Configuring Voice Services
Secure Call Implementation
Secure Call Implementation
This section describes secure call implementation with the ATA device . It includes the following topics:
• “Enabling Secure Calls” section on page 71
• “Secure Call Details” section on page 72
• “Using a Mini-Certificate” section on page 73
• “Generating a Mini Certificate” section on page 73
4
NOTE This is an advanced topic meant for experience installers. See also the
Provisioning Guide
.
Enabling Secure Calls
A secure call is established in two stages. The first stage is no different from normal call setup. The second stage starts after the call is established in the normal way with both sides ready to stream RTP packets.
In the second stage, the two parties exchange information to determine if the current call can switch over to the secure mode. The information is transported by base64 encoding embedded in the message body of SIP INFO requests, and responses using a proprietary format. If the second stage is successful, the ATA device plays a special Secure Call Indication Tone for a short time to indicate to both parties that the call is secured and that RTP traffic in both directions is being encrypted.
If the user has a phone that supports call waiting caller ID (CIDCW) and that service is enabled, the CID will be updated with the information extracted from the Mini-Certificate received from the remote party. The Name field of the CID will be prepended with a ‘$’ symbol. Both parties can verify the name and number to ensure the identity of the remote party.
LV S
The signing agent is implicit and must be the same for all ATAs that communicate securely with each other. The public key of the signing agent is pre-configured into the ATA device by the administrator and is used by the ATA device to verify the Mini-Certificate of its peer. The Mini-Certificate is valid if it has not expired, and it has a valid signature.
Page 72
Configuring Voice Services
Secure Call Implementation
The ATA device can be configured so that, by default, all outbound calls are either secure or not secure. If secure by default, the user has the option to disable security when making a call by dialing *19 before dialing the target number. If not secure by default, the user can make a secure outbound call by dialing *18 before dialing the target number. However, the user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.
The ATA device will not switch to secure mode if the CID of the called party from its Mini-Certificate does not agree with the user-id used in making the outbound call. The ATA device performs this check after receiving the Mini-Certificate of the called party
Secure Call Details
Looking at the second stage of setting up a secure call in greater detail, this stage can be further divided into two steps.
4
STEP 1 The caller sends a “Caller Hello” message (base64 encoded and embedded in the
message body of a SIP INFO request) to the called party with the following information:
• Message ID (4B)
• Version and flags (4B)
• SSRC of the encrypted stream (4B)
• Mini-Certificate (252B)
Upon receiving the Caller Hello, the called party responds with a Callee Hello message (base64 encoded and embedded in the message body of a SIP response to the caller’s INFO request) with similar information, if the Caller Hello message is valid. The caller then examines the Callee Hello and proceeds to the next step if the message is valid.
STEP 2 The caller sends the “Caller Final” message to the called party with the following
information:
• Message ID (4B)
• Encrypted Master Key (16B or 128b)
• Encrypted Master Salt (16B or 128b)
Page 73
Configuring Voice Services
Secure Call Implementation
Using a Mini-Certificate
The Master Key and Master Salt are encrypted with the public key from the called party mini-certificate. The Master Key and Master Salt are used by both ends for deriving session keys to encrypt subsequent RTP packets. The called party then responds with a Callee Final message (which is an empty message).
The Mini-Certificate (MC) contains the following information:
• User Name (32B)
• User ID or Phone Number (16B)
• Expiration Date (12B)
• Public Key (512b or 64B)
• Signature (1024b or 512B)
4
The MC has a 512-bit public key used for establishing secure calls. The administrator must provision each subscriber of the secure call service with an MC and the corresponding 512-bit private key. The MC is signed with a 1024-bit private key of the service provider, which acts as the CA of the MC. The 1024-bit public key of the CA signing the MC must also be provisioned for each subscriber.
The CA public key is used to verify the MC received from the other end. If the MC is invalid, the call will not switch to secure mode. The MC and the 1024-bit CA
and
Mini
public key are concatenated and base64 encoded into the single parameter
Certificate
parameter, which should be kept secret, like a password. (
SRTP Private Key
Because the secure call establishment relies on exchange of information embedded in message bodies of SIP INFO requests/responses, the service provider must ensure that the network infrastructure allows the SIP INFO messages to pass through with the message body unmodified.
. The 512-bit private key is base64 encoded into the
Mini Certificate
are configured in the Line tabs.)
SRTP Private Key
Generating a Mini Certificate
Cisco provides a Mini Certificate Generator for the generation of mini certificates and private keys. Partners can download the Mini Certificate Generator by going to Cisco Partner Central, Voice & Conferencing page, Technical Resources section. Use the following URL:
http://www.cisco.com/web/partners/sell/smb/products/ voice_and_conferencing.html#~vc_technical_resources
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Configuring Voice Services
Secure Call Implementation
NOTE The partner sites require a logon.
The Mini Certificate Generator uses the following syntax:
gen_mc ca-key user-name user-id expire-date
Where:
• ca-key is a text file with the base64 encoded 1024-bit CA private/public key
9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qq e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZ YTccnZ75TuTjj13qvYs=5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/ Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/ IqSrsf6scsmundY5j7Z5mK5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3M F+zjyYrVUFdM+pXtDBxmM+fGUfrpAuXb7/k=
4
pairs for signing/verifying the MC, such as the following:
• user-name is the name of the subscriber, such as “Joe Smith”. Maximum length
is 32 characters
• user-id is the User ID of the subscriber, which must match exactly the user-id
used in the INVITE when making the call, such as “14083331234”. The maximum length is 16 characters.
• expire-date is the expiration date of the MC, such as “00:00:00 1/1/34”
(34=2034). Internally the date is encoded as a fixed 12B string: 000000010134
The tool generates the
Mini Certificate
be provisioned.
EXAMPLE:
gen_mc ca_key “Joe Smith” 14085551234 “00:00:00 1/1/34”
This example produces the following Mini Certificate and SRTP Private Key:
<Mini Certificate> Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEw MTM00OvJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/ xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhx ES767G0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/ uQ/ LJQlni2kwqlm7scDvll5k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3AmcqE 9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBx uMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs= <SRTP Private Key> b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/ e+Klm4XiiFEPaKmU8UbooxKG36SEdKusp0AQ==
and
SRTP Private Key
parameters that can
Page 75
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
SIP Trunking and Hunt Groups on the SPA8000
The SPA8000 supports SIP Trunking, which allows you to connect a traditional PBX to VoIP services. In this configuration, calls go through the ITSP rather than the PSTN, yet the call routing functionality is similar to that of traditional PSTN lines.
You can configure up to four trunk groups for the purpose of inbound call routing and outbound caller identification. You can configure a trunk number on the SPA8000, such that an incoming call automatically rings the grouped lines simultaneously or in a specified order. For outbound calls, SIP Trunking ensures that all calls on a trunk line can be identified by the trunk number and a common caller ID. This feature helps you to ensure that calls are directed to available lines and that work groups such as sales teams can work together to answer calls. In addition, teams can project a common identity when placing outbound calls on a trunk.
4
This section provides information about SIP trunking and explains how to configure your trunk groups.
Refer to the following topics:
• “About SIP Trunking,” on page 76
• “Setting the Trunk Group Call Capacity,” on page 78
• “Inbound Call Routing for a Trunk Group,” on page 78
• “Contact List for a Trunk Group,” on page 79
• “Outgoing Call Routing for a Trunk Group,” on page 81
• “Configuring a Trunk Group,” on page 82
• “Additional Notes About Trunk Groups,” on page 85
• “Setting the Hunt Policy,” on page 85
• “Trunk Group Management,” on page 83
Page 76
Configuring Voice Services
InternetIntegrated
Access Device
SPA8000
ITSP
Fax
PBX System
Fax
PBX System
194488
SIP Trunking and Hunt Groups on the SPA8000
About SIP Trunking
The SIP Trunking feature allows a traditional PBX to seamlessly migrate from PSTN service to VoIP service over a broadband link. The SPA8000 offers up to eight telephone lines to the PBX.
4
The SPA8000 offers four trunk groups, numbered T1, T2, T3, and T4. A SIP-based voice service with an ITSP can be configured on each trunk group with a distinct phone number. Each of the eight SPA8000 lines can be configured either as a standalone line, as in a classic ATA FXS port, or as a trunk line that is associated with a trunk group.
• Inbound calling: A trunk group offers a single number for callers to call into the
small business, with the capability to programmatically ring one or more trunk lines.
• Outbound calling: When a PBX phone makes a call, the PBX selects one of the
available trunk lines. The trunk line assumes the Caller ID of the trunk group.
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Configuring Voice Services
Phone 1
Phone 2
Phone 3
Phone 4
Phone 5
Phone 6
Phone 7
Phone 8
L1
L2
L3
L4
L5
L6
L7
L8
T1
T2
T3
T4
Internal
RTP Path
SIP Path
ITSP
Proxy
Server
SIP Trunking and Hunt Groups on the SPA8000
The following figure shows a simplified logical block diagram of the SPA8000 with the SIP Trunking feature.
Figure1 Logical Block Diagram of SIP Trunking
4
• SIP Path: As a standalone line, the SIP User Agent (SIP UA) exchanges signaling
directly with the ITSP equipment. As a trunk line, the Line UA exchanges signaling with the internal proxy server only. The Internal Proxy Server handles all SIP signalling between both ends of the call, from call establishment to termination.
• RTP Path: Whether the line is standalone or a member of a trunk group, the Line
UA exchanges RTP packets directly with the ITSP equipment.
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
NOTE Although the figure shows only one ITSP account, each standalone line and each
Trunk Group can be configured with a different ITSP (with some limitations applied).
Setting the Trunk Group Call Capacity
The ITSP may set a limit to the number of calls that can be made on a trunk group. You can configure a trunk group’s call capacity parameter to meet the requirements of the ITSP. Both incoming call and outgoing calls are counted towards this limit. The call capacity has the following impact on call handling:
• Inbound calls: When the limit is reached, the Trunk UA replies 486 to the
caller.
• Outbound calls: When the limit is reached, the Line UA plays a fast busy
tone to the caller. Note that a trunk line can make an outgoing call only through its own trunk. If that trunk reaches full capacity, it will not attempt to failover to use other trunks.
4
You can configure this setting in the Voice tab > Trunk (T1 ... T4) page, Subscriber Information section, Call Capacity field. For more information, see “Configuring a
Trunk Group,” on page 82.
Inbound Call Routing for a Trunk Group
An incoming call is handled as follows:
STEP 1 When an incoming call is detected by the Trunk UA, the UA first checks if there is
capacity to handle the call. If there is insufficient capacity, the UA rejects the call with a 486 response.
STEP 2 If there is spare call capacity, the UA consults the Contact List to determine which
line or lines to ring (that is, for the proxy to send SIP INVITE to), and starts “hunting.” (See “Configuring a Trunk Group,” on page 82)
STEP 3 When a line is selected to ring, one or more PBX phones may be alerted, according
to the PBX features and configuration.
STEP 4 The Caller ID of the external Caller is signaled by the Line UA out to the FXS port
using the configured Caller ID method (FSK, DTMF, etc.). The PBX must be able to detect Caller ID signal in order for the proper Caller ID to show.
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
STEP 5 If the call is picked up by the PBX, the Line UA replies 200 OK with SDP to the
internal Proxy. The Trunk UA in turn replies 200 OK to the ITSP and relay the Line SDP in the 200 OK message also. If all goes well, the Line UA and the ITSP equipment start exchanging RTP packets afterwards.
Contact List for a Trunk Group
The hunting process for incoming calls is controlled by the Contact List. The Contact List specifies the lines to ring, the order in which to ring them, the duration to ring one line before trying another line, and the maximum period to hunt. Below, the syntax is described and examples are provided to help you to configure the Contact List for each trunk group.
SYNTAX: line[,line[,line[…]]],hunt=hrule[,cfwd=target]
4
• line: The line numbers (1 - 8), or a wildcard * or ? to represent all lines.
• The Trunk UA rings only trunk lines, that is, lines that are assigned to a trunk
group through the Voice tab > Line page, Tr unk Grou p field. The Trunk UA does not ring any standalone lines that are included in the Contact List. The Trunk UA rings any trunk line that is included in the list, even if it is not assigned to the particular trunk group for this Contact List.
• You can instruct the SPA8000 to hunt only the phones that are on-hook,
through the Voice tab > SIP page, Trunking Parameters section, Hunt Policy field. See “Setting the Hunt Policy,” on page 85.
• hunt=hrule: The hunt order, ring interval, and maximum duration, in the
following format: hunt =algo;interval;max
• algo: The hunt order.
- re: Restart. Hunting starts at the beginning of the list. If the first line does
not answer within the specified interval (see below), the hunt proceeds through the lines in sequential order.
- ne: Next. The Trunk UA determines the line that was chosen in the
previous hunt, and hunting starts with the next line in the list. If that line does not answer within the specified interval (see below), the hunt proceeds through the lines in sequential order.
- ra: Random order. The Trunk UA randomly chooses a line from the list. If
the selected line does not answer within the specified interval (see
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
below), the hunt proceeds randomly through the unchosen lines until each line is tried.
- al: All. The Trunk UA rings all the lines at the same time.
• interval: The number of seconds to wait for one line to answer, before
choosing another line. If interval is *, the hunt is stopped at the first line that starts ringing, and rings the line until it answers, or the caller hangs up, or the line's ringer times out.
• max: The maximum duration of the hunt, either in seconds or cycles. When
this limit is reached, the call is rejected or is forwarded to the specified call forward number (see below).
- If max is greater than interval, it represents the total time in seconds
to hunt.
- If max is less than interval, it represents the maximum number of
times to cycle through the hunt group. If max is 0, hunting continues indefinitely until the caller either hangs up or the call is answered. Exceptions: This value is ignored if algo = all, or interval = * (but it must be present and should be set to 1).
4
• cfwd=target: If the call is unanswered and the maximum hunting duration
has been met, the call is forwarded to the specified number. When forwarding the call, the SPA8000 sends a 302 response to the ITSP.
NOTE The call forward settings for the individual lines are ignored during hunting. Instead,
the cfwd settings in the Contact List are used.
EXAMPLES:
• 1,2,3,4,5,6,7,8,hunt=re;*;1
Lines 1 through 8 are included (1,2,3,4,5,6,7,8). The hunt starts at the beginning of the list (hunt=re). When an available line is found, the call stays with the line until the call is either answered, rejected, or cancelled by the caller (* is entered for interval).
• ?,hunt=al;30;1,cfwd=14085550100
A wildcard character (?) is used to represent “all trunk lines.” All lines ring simultaneously (hunt=al). If there is no answer after 30 seconds (30), the call is forwarded to the specified number (cfwd=14085550100).
• ?,hunt=ra;12;1,cfwd=14085550123
A wildcard character is used to represent “all trunk lines.” The Trunk UA
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
chooses lines in random order (hunt=ra). If a selected line does not answer within 12 seconds (12), the Trunk UA chooses another line at random. If there is no answer after 1 cycle (1), the call is forwarded to forwarded to the specified number (cfwd=14085550123).
• ?,hunt=ra;*;1,cfwd=14085550155
A wildcard character is used to represent “all trunk lines.” The Trunk UA chooses lines in random order (hunt=ra). The interval is *, meaning the hunt stops when a selected line starts ringing, and will ring the line until it answers, or the caller hangs up, or the line's ringer times out. If the ringer times out, the call is automatically forwarded to the specified number (cfwd=14085550155).
Outgoing Call Routing for a Trunk Group
Outbound calls on a trunk line are handled as follows:
4
STEP 1 When a PBX phone selects an outside line, the PBX looks for an open line. If the
PBX finds an open line, it takes the line off hook and bridges the audio between the PBX phone and the line. On detecting the off hook signal, the SPA8000 Line UA plays dial tone and ready to collect digits from the PBX phone.
STEP 2 As the PBX phone user dials the number, the Line UA applies its dial plan to the
number. If the Line UA detects an invalid number, it rejects the all by playing reorder tone, then howling tone, then silence. If a valid number is received, it sends a SIP INVITE message to the internal Proxy.
STEP 3 The Proxy routes the call to the trunk group UA for the line, and the trunk group UA
will attempt to place the call to the ITSP if there is available capacity on the trunk. If there is no call capacity left on the trunk, the internal Proxy will reject the INVITE from the Line UA, which in turn terminates the call and plays reorder tone out to the FXS port.
NOTE The SPA8000 will also apply the Trunk Dial Plan on the number before sending out
INVITE to the ITSP. This Trunk Dial Plan typically is redundant since the trunk should trust the number sent by the Line UA. By default the trunk dial plan allows any non­empty number: ([*#0-9A-D][*#0-9A-D].)
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
Configuring a Trunk Group
To configure a hunt group, you must first specify the trunk lines by assigning lines to trunk groups. Then you enter the account information, specify the call capacity, and configure the Contact List.
Before you begin this procedure, determine which lines you want to associate with each trunk group that you are configuring. Refer to the following example:
Line Trunk Group
1, 3, 5 T1
4, 6, 8 T2
2None
4
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Assign each line to a trunk group, as needed:
a. Click Voice tab > L
b. In the Trunk Gr oup field, near the top of the line configuration page, choose a
trunk number or choose none for a standalone line (the default setting).
c. Repeat this step for each line that you want to add to a trunk group.
n
, where n represents the number of the line interface.
Voice > Ln > Trunk Group field
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
STEP 3 Enter the settings for each trunk group, as needed:
a. Click Voice tab > T
b. Enter the account information in the Subscriber Information section.
• Display Name: The Caller ID that you want to use for outbound calls on this
line
• User ID: Your account number with the ITSP (usually the telephone number)
• Password: Your password for this ITSP account
c. In the Call Capacity field, enter the maximum number of concurrent calls
allowed by your ITSP, or leave the default setting, unlimited (16 calls).
d. In the Contact List field, modify the contact list as needed. See “Contact List
for a Trunk Group,” on page 79.
n
, where n represents the trunk group number (T1 ... T4).
4
e. Repeat this step for each trunk group that you need to configure.
STEP 4 Click Submit All Changes.
Trunk Group Management
You can check the status of the trunks by clicking the Trunk Status link, which appears both at the top right corner of the web page and at the lower left corner.
You also can connect directly to the Trunk Status Page by entering the following URL: http://spa8000-ip-addr/status. This page is available with the User Login or the Admin Login.
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Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
The Trunk S ta tu s page shows all calls that are currently active on each trunk group.
This page shows a snapshot of the trunk activity. You can refresh the data at any time by clicking the Refresh button on the web browser toolbar. The page shows the following information:
4
Trunk Status page
• External: The called number
• Station: The SPA8000 line that is in use for this call
• Direction: The direction of the call, either Outbound or Inbound
• State: The state of the call
• Calling: An outbound call was initiated but is not ringing at the other end.
• Proceeding: The outbound call is ringing at the other end.
• Ringing: An inbound call is ringing.
• Connected: The call is connected.
• Duration: The duration of the call
In the case of a hung call, you can select the check box for the call and then click the Delete button to cancel the call.
Page 85
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
Setting the Hunt Policy
You can configure the SPA8000 so that the hunt rule applies to all phone or only to the phones that are on hook.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > SIP.
STEP 3 Scroll down to the Trunkin g Par amet er s section.
STEP 4 In the Hunt Policy field, choose the desired option:
• onhook only: The hunt includes only the phones that are on hook.
• any state: The hunt includes all phones regardless of the state.
4
STEP 5 Click Submit All Changes.
Additional Notes About Trunk Groups
This section includes information about other topics that may be of interest when you are configuring trunk groups:
• Voice mail: There is no individual mail box for a trunk line. For example, if lines 1,
2, 3, and 4 belong the trunk group T1, then the four lines implicitly share the same voice mail box from the ITSP. When there is new voice mail waiting in the trunk mail box, the UAs for all four lines will be notified by the ITSP via the internal Proxy, and all four lines will show the message waiting indicator, such as by playing stutter dial tone, if enabled by the administrator.
• Supplementary features: Supplementary features are offered at the line level
only, not at the trunk level. Via the PBX, the phone user can trigger/control supplementary service and settings by signaling to the line port or configuring the line parameters. For more information, refer to the Appendix B, “ATA Voice
Field Reference.”
Page 86
Configuring Music on Hold
This chapter explains how to configure Music on Hold using either a music file or streaming audio.
This chapter includes the following topics:
• “Using the Internal Music Source for Music On Hold,” on page 86
• “Configuring a Streaming Audio Server,” on page 88
5
Using the Internal Music Source for Music On Hold
An internal music source with the user ID imusic is available. It plays an internally stored music file repeatedly. The unit ships with a default music file (
Amor
). You can override this file by downloading a new file into the unit by using
TFTP.
Refer to the following topics:
• “Using the Internal Music Source,” on page 86
• “Changing the Music File for the Internal Music Source,” on page 87
Using the Internal Music Source
To use the internal music source, simply identify imusic as the MOH server for each IP phone.
STEP 1 Use the phone menu to find the IP address of the phone:
a. Press the Setup button on the phone keypad.
Romance de
b. Press 9 - Network, and then scroll down to 2- Current IP Address.
Page 87
Configuring Music on Hold
Using the Internal Music Source for Music On Hold
STEP 2 Start Internet Explorer, and then enter the IP address of the telephone. The
Telephone Configuration page appears in a separate browser window.
STEP 3 Click Admin Login, and then click Advanced.
STEP 4 Click the Ext 1 tab.
5
STEP 5 Scroll down to the
STEP 6 Enter the following value in the
STEP 7 Click Submit All Changes.
STEP 8 To verify, place a test call to the extension. When the call is answered and put on
hold, the caller should hear the default music file (
Call Feature Settings
MOH Server
section.
field: imusic
Romance de Amor
).
Changing the Music File for the Internal Music Source
The following resources are required to change the music file for the internal music source:
• TFTP server software
• The IP address of the administration computer that is connected to the
SPA9000
• A music source in G.711u format, sampled at 8000 samples/sec, up to 65.5
seconds in length, with no header information
STEP 1 Before you begin, make sure that you have TFTP server software running on your
computer.
STEP 2 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
STEP 3 Click Voice tab > SIP.
STEP 4 Scroll down to the
STEP 5 Enter the following URL in the Internal Music URL field:
tftp://server_IPaddress:portpath
• server_IPaddress: The local IP address of the computer you are using as the
TFTP server
• port: The port number used by the TFTP server (default 69)
Internal Music Source Parameters
section.
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Configuring Music on Hold
Configuring a Streaming Audio Server
• path: The location and name of a music file in the correct format
• For example, if the computer local IP address is 192.168.0.5, the directory is
named would enter the following URL: tftp://192.168.0.5:69/musicdir/
jazzmusic.dat
STEP 6 Click Submit All Changes. The unit reboots. Then the unit downloads the file and
stores it in flash memory.
musicdir
, and the converted music file is named
Configuring a Streaming Audio Server
This section describes how to use and configure a streaming audio server (SAS). It includes the following topics:
jazzmusic.dat
5
, then you
• “About the Streaming Audio Server,” on page 88
• “Configuring the Streaming Audio Server,” on page 90
• “Using the IVR with an SAS Line,” on page 91
About the Streaming Audio Server
The Streaming Audio Server (SAS) feature lets you attach an audio source to an FXS port and use it as a streaming audio source device. If the unit has multiple FXS ports, either or both of the associated lines can be configured as an SAS server.
Use a media signal adapter or “music coupler” to connect an Ethernet cable from a media source to the FXS port. For example, the MC-9700 Music Coupler has been tested with ATA devices and is available at the following URL:
www.neogadgets.com/cart/ cart.php?target=product&product_id=17&substring=music+coupler
Page 89
Configuring Music on Hold
Configuring a Streaming Audio Server
After you complete the required configuration, the FXS port is ready to stream audio. The functionality depends on the hook state of the FXS port:
• If the FXS port is off hook, an incoming call is answered automatically and
audio is streamed to the calling party.
NOTE Each SAS server can maintain up to five simultaneous calls. If the
• If the FXS port is on-hook when the incoming call arrives, a SIP 503 response
code is transmitted to indicate “Service Not Available.”
• If an incoming call is auto-answered, but later the FXS port changes to on-hook,
the call is not terminated but continues to stream silence packets to the caller.
5
second line on the unit is disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls receive a busy signal (SIP 486 Response).
• The SAS line can be set up to refresh each streaming audio session
periodically using a SIP re-INVITE message, which detects if the connection to the caller is down. If the caller does not respond to the refresh message, the SAS line terminates the call so that the streaming resource can be used for other callers.
Additional information:
• The SAS line does not ring for incoming calls even if the attached equipment is
on-hook.
• If no calls are in session, battery is removed from tip-and-ring of the FXS port.
Some audio source devices have an LED to indicate the battery status. This can be used as a visual indication as to whether audio streaming is in progress.
• Call Forwarding, Call Screening, Call Blocking, DND, and Caller-ID Delivery
features are not available on an SAS line.
Page 90
Configuring Music on Hold
Configuring a Streaming Audio Server
Configuring the Streaming Audio Server
Use the following procedure to configure an SAS with an external music source.
STEP 1 Connect an RJ-11 adapter between the music source (a CD player or iPod, for
example) and an FXS port.
STEP 2 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
STEP 3 Configure the FXS port:
5
a. Click Voice tab > FXS
where you connected the cable from the external music source.
b. In the Subscriber Infomation section, enter the following settings:
• Display Name: Enter an extension number of name for the FXS 1 port, such
as Receptionist Area Fax Machine.
• User ID: Enter a three- to four-digit extension number that is not is use by
another extension.
c. In the Streaming Audio Server (SAS) section, choose yes from the SAS
Enable drop-down list.
STEP 4 Click Submit All Changes.
STEP 5 Configure each phone to use this audio source as the MOH server:
a. Click the PBX Status link to view the list of phones.
b. In the list, find the phone that you want to configure, and then click the hyperlink
in the IP Address column. The Telephone Configuration page appears in a separate window.
N
, where N represents the number of the FXS port
c. Click the Ext 1 tab.
d. Scroll down to the Call Feature Settings section.
e. In the MOH Server field, enter the extension number that you assigned to the
FXS port for the streaming audio server.
f. Click Submit All Changes.
Page 91
Configuring Music on Hold
Configuring a Streaming Audio Server
g. Close the window for the Telephone Configuration page.
h. Repeat this step to configure each phone, as needed.
Using the IVR with an SAS Line
The IVR can still be used on an SAS line, but the user needs to follow the following steps:
STEP 1 Power off the ATA device.
STEP 2 Connect a phone to the port and make sure the phone is on-hook.
STEP 3 Power on the ATA device.
5
STEP 4 Pick up handset and press * * * * to invoke IVR in the usual way.
If the ATA device boots and finds that the SAS line is on-hook, it does not remove battery from the line so that IVR may be used. But if the ATA device boots up and finds that the SAS line is off-hook, it removes battery from the line because no audio session is in progress.
Page 92
6
Configuring the PSTN (FXO) Gateway on the SPA3102
This chapter describes how to configure the PSTN gateway on the SPA3102 and the SPA8800.
• “Connecting to PSTN and VoIP Services” section on page 92
• “How VoIP-To-PSTN Calls Work” section on page 93
• “How PSTN-To-VoIP Calls Work” section on page 95
• “Configuring VoIP Failover to PSTN” section on page 98
• “Sharing One VoIP Account Between the FXS and PSTN Lines (SPA3102)”
section on page 98
• “Other Options” section on page 99
• “Call Scenarios” section on page 100
Connecting to PSTN and VoIP Services
Both the SPA3102 and the SPA8800 allow your analog and IP phones to participate in calls over the VoIP network and the PSTN. Both devices provide a VoIP-To-PSTN calling function, which is referred to as a PSTN-To-VoIP calling function, which is referred to as a
These ATAs function somewhat differently because they are designed to meet different business needs.
PSTN gateway
VoIP gateway
, as well as a
.
• The SPA3102 is designed to work independently as a PSTN gateway and
VoIP gateway, without relying on a PBX for call control.
- FXS Port: The SPA3102 has 1 FXS port that you can connect to a
standard analog telephone or fax machine. Configure the FXS settings by using the SPA3102 Line page.
Page 93
Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
- FXO Port: The SPA3102 has 1 FXO port that you can connect to the
PSTN. Configure the FXO settings by using the SPA3102 PSTN Line page.
• The SPA8800 is designed to work with your PBX as a PSTN gateway and a
VoIP gateway. Call control is provided by a standard PBX, an Asterisk­based IP PBX, the SPA9000 Voice System, or an Internet-based call control system.
- FXS Port: The SPA8800 has 4 FXS ports that you can connect to
standard analog telephones or fax machines. Configure the FXS settings by using the SPA8800 Phone 1-4 pages.
- FXO Port: The SPA8800 has 4 FXO port sthat you can connect to the
PSTN. Configure the FXO settings by using the SPA8800 Line 1-4 pages.
6
How VoIP-To-PSTN Calls Work
To obtain PSTN services through the SPA3102 or the SPA8800, the VoIP caller establishes a connection with the PSTN Line by way of a standard SIP INVITE request addressed to the PSTN Line.
One-Stage Dialing (SPA3102 and SPA8800)
One-stage dialing allows a call to be started over VoIP and then immediately get a dial tone on the PSTN. When you take a phone off hook and dial a number, the call is automatically routed to the VoIP or the PSTN, based on the dial plan.
On the SPA3102, you can enable HTTP Digest Authentication. In this case, the SPA3102 challenges the INVITE with a 401 response if it does not have a valid Authorization header. The Authorization header should include a < parameter, where n refers to one of eight VoIP user accounts that can be configured on the ATA device. The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The < parameter must match one of the VoIP accounts stored on the ATA device. You can configure these settings on the Voice tab > PSTN Line page. For more information, see “VoIP Users and Passwords (HTTP Authentication) section,” on page 213.
User ID
User ID
n>
n>
Page 94
Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
Two-Stage Dialing (SPA3102)
In two-stage dialing, the SPA3102 takes the FXO port off-hook but does not automatically dial any digits after accepting the call. To invoke two-stage dialing, the VoIP caller should INVITE the PSTN Line without the user-id in the Request-URI or with a user-id that matches exactly the < user-id in the Request-URI is treated as a request for one-stage dialing if one­stage dialing is enabled, or dropped by the SPA3102 (as if no user-id is given) if one-stage dialing is disabled.
HTTP Digest Authentication can be also used for two-stage dialing, as in one­stage dialing. If using HTTP Digest Authentication or Authentication is disabled, the VoIP caller should hear the PSTN dial tone right after the call is answered (by a SIP 200 response).
You also can enable PIN authentication. In this case, the VoIP caller is prompted to enter a PIN number after the SPA3102 answers the call. The PIN number must end with a # key. The inter-PIN-digit timeout is 10 seconds (not configurable). Up to eight VoIP caller PIN numbers can be configured on the SPA3102. A dial plan can be selected for each PIN number. If the caller enters a wrong PIN or the SPA3102 times out waiting for more PIN digits, the SPA3102 tears down the call immediately with a BYE request.
User ID
6
n> of the PSTN Line. A different
The call scenarios may involve the following types of callers:
• VoIP caller—Someone who calls the ATA device via VoIP to obtain PSTN
service
• VoIP user—A VoIP caller that has a user account (user-id and password) on the
SPA3102
• PSTN caller—Someone who calls the ATA device from the PSTN to obtain VoIP
service
VoIP callers can be authenticated by one of the following methods:
• No Authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
• HTTP digest—SIP INVITE must contain a valid authorization header.
PSTN callers can be authenticated by one of the following methods:
• No authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
Page 95
Configuring the PSTN (FXO) Gateway on the SPA3102
How PSTN-To-VoIP Calls Work
NOTE When the source address of the INVITE is 127.0.0.1, authentication is automatically
disabled because this is a call by the local user. This applies to both one-stage and two-stage dialing.
These settings can be configured on the SPA3102 PSTN Line page. See “VoIP-
To-PSTN Gateway Setup section,” on page 211.
How PSTN-To-VoIP Calls Work
For PSTN-to-VoIP calls, the basic PSTN-to-VoIP call flow is as follows:
1. When a PSTN call comes in to the ATA device and is unanswered (after a configurable number of rings), then the ATA device takes the FXO port off hook.
6
2. The ATA device plays dial tone.
3. The PSTN caller enters the target telephone number. The collected digits are processed by the default dial plan.
On the SPA3102, you can add PIN authentication to the basic flow:
1. When a PSTN call comes in to the ATA device and is unanswered (after a configurable number of rings), then the ATA device takes the FXO port off hook.
2. The SPA3102 prompts the caller to enter the PIN number followed by the # key.
3. The SPA3102 compares the PIN to the configured PSTN PIN values.
• If the PIN matches one of the configured PSTN PIN values, then the
SPA3102 plays dial tone. The caller enters the telephone number and the collected digits are processed by the dial plan associated with the PIN number. (These dial plans are configured on the Voice Voice tab > PSTN Line page, Dial Plans section.)
• If the PIN does not match one of the configured PSTN PIN values, then the
ATA device plays the reorder tone and then takes the FXO port on-hook.
Page 96
Configuring the PSTN (FXO) Gateway on the SPA3102
How PSTN-To-VoIP Calls Work
NOTE For information about configuring the timer values for the above scenarios, see
“FXO (PSTN) Timer Values (sec) section,” on page 214.
For information about configuring caller authentication on the SPA3102, see “VoIP-
To-PSTN Gateway Setup section,” on page 211.
Terminating Gateway Calls
There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call leg. A gateway call is terminated when either call leg is ended. When the call terminates, the FXO port goes on-hook so the PSTN line is available for use. The ATA device detects that the PSTN call leg is ended when one of the following conditions occurs during a call:
6
• The PSTN Line voltage drops to a very low value (this occurs if the line is
disconnected from the PSTN service or if the PSTN switch provides a CPC signal).
• A polarity reversal or disconnect tone is detected at the FXO port.
• There is no voice activity for a configurable period of time in either direction
at the FXO port.
When any of the above conditions occur, the ATA device takes the FXO port on hook and sends a BYE request to end the VoIP call leg. On the other hand, when the ATA device receives a SIP BYE from the VoIP during a call, it takes the FXO port on hook to end the PSTN call leg.
In addition, the ATA device can also send a refresh signal periodically to the VoIP call leg to determine whether the call leg is still up. If a refresh operation fails, the ATA device ends both call legs.
On the SPA3102, these settings can be configured on the Voice tab > PSTN Line page. For more information, see “PSTN Disconnect Detection section,” on
page 216.
On the SPA8800, these settings can be configured on the Voice tab > Phone page. For more information, see “FXS Port Polarity Configuration section,” on
page 183.
Page 97
Configuring the PSTN (FXO) Gateway on the SPA3102
VoIP Outbound Call Routing (SPA3102)
VoIP Outbound Call Routing (SPA3102)
On the SPA3102, calls made from Line 1 are routed through the configured Line 1 service provider, by default. You can override this behavior by IP dialing, through which the calls can be routed to any IP address that the user enters. The SPA3102 allows flexible call routing with four sets of gateway parameters and configurable dial plans. These settings are configured on the SPA3102 Voice tab > Line page. See “Gateway Accounts section (SPA3102),” on page 180.
NOTE The PBX handles outbound call routing for the SPA8800.
You can specify Gateways 1 to 4 in a dial plan by using the identifiers gw1, gw2, gw3, or gw4. Also, gw0 represents the internal PSTN gateway via the FXO port. You can specify in the dial plan to use gw In general, you can specify any gateway address in the dial plan. In addition, three parameters are added that can be used with call routing:
x (x
6
= 0,1,2,3,4) when making certain calls.
• usr: User-id used for authentication with the given gateway
• pwd: Password used for authentication with the given gateway
• nat: Enable or disable NAT mapping when calling the gateway
The following table lists some examples.
Example Description
<9,:>xx.<:@gw1
[93]11<:@gw0>
<8,:1408>xxxxxxx<:@pstn. cisco.com:5061;usr=joe; pwd=joe_pwd;nat>
<8,:1408>xxxxxxx<:@gw2:5061; usr=”Alex Bell”;pwd= ”anything”;nat=no>
Dial 9 to start outside dial tone, followed by one or more digits, and route the call to Gateway 1.
Route 911 and 311 calls to the local PSTN gateway
Dial 8 to start outside dial tone, prepend 1408 followed by seven digits, and route the call to pstn.cisco.com:5061, with user-id = joe, and pwd = bell_pwd, and enable NAT mapping
Dial 8 to start outside dial tone, prepend 1408 followed by seven digits, and route the call to Gateway 2, but use the given port, user-id, and password, and no pstn.cisco.com:5061, and with user-id = “Alex Bell” and pwd = bell_pwd, and disable NAT mapping
Page 98
Configuring the PSTN (FXO) Gateway on the SPA3102
Configuring VoIP Failover to PSTN
You can set up multiple PSTN gateways at different locations and configure Line 1 to use a different gateway when dialing specific numbers.
Configuring VoIP Failover to PSTN
When power is disconnected from the ATA, the FXS port is connected to the FXO port. In this case, the telephone attached to the FXS port is electrically connected to the PSTN service via the FXO port. When power is applied to the ATA device, the FXS port is disconnected from the FXO port. However, if the PSTN line is in use when the power is applied to the ATA device, the relay is not flipped until the PSTN line is released. This feature ensures that the ATA device does not interrupt any call in progress on the PSTN line.
When Line 1 VoIP service is down (because of registration failure or loss of network link), the ATA device can be configured to automatically route all outbound calls to the internal gateway using the parameter listed below.
6
For SPA3102, you can configure this setting on the Voice tab > Line page. For SPA8800, you can configure this setting on the Voice tab > Phone page. For more information, see “VoIP Fallback to PSTN section (SPA3102 and SPA8800),” on
page 180.
Sharing One VoIP Account Between the FXS and PSTN Lines (SPA3102)
On the SPA3102, both the FXS (Line 1) and FXO (PSTN Line) can receive incoming calls for a single VoIP account if they are different ports. Consider the following points:
• If the service provider allows multiple registration contacts and
simultaneous ringing, both lines can register periodically with the service provider. In this case, both lines receive inbound calls to this VoIP account. The PSTN Line should be configured with a sufficiently long answer delay before the call is automatically answered to allow for the function of the PSTN gateway.
• If the service provider does not allow more than one register contact, the
PSTN Line should not register. In this case, only Line 1 rings on the inbound call to this VoIP account because it is the only line registered with the service provider.
Page 99
Configuring the PSTN (FXO) Gateway on the SPA3102
Other Options
• Line 1 can have the call forwarded to the PSTN Line after a few seconds
using the Call-Forward-On-No-Answer feature with gw0 as the forward destination. Similarly, Line 1 can apply Call-Forward-All, Call-Forward-On­Busy, and Call-Forward-Selective feature, and direct the caller to the PSTN­Gateway.
• Only PIN authentication is allowed when a VoIP caller is forwarded to the
PSTN-gateway from Line 1. If HTTP Authentication is used, the caller is not authenticated.
• When using the Forward-To-GW0 feature, you can forward the caller to a
specific PSTN number, using the syntax < forward destination. When using this with Call-Forward-Selective, you can develop some interesting applications. For example, you can forward all callers with 408 area code to 14081234567, or all callers with 800 area code to 18005558355 (This is the number for Tell Me). When this syntax is used, authentication is not used and the target PSTN number is automatically dialed by the ATA device after the caller is forwarded to gw0.
PSTN-number
6
>@gw0 in the
Other Options
This section describes other options provided by the SPA3102 and the SPA8800:
• “PSTN Call to Ring Line 1 (SPA3102)” section on page 99
• “Symmetric RTP (SPA3102 and SPA8800)” section on page 99
• “Call Progress Tones (SPA3102 and SPA8800)” section on page 100
PSTN Call to Ring Line 1 (SPA3102)
On the SPA3102, this feature allows a PSTN caller to ring Line 1. When the PSTN line rings, the PSTN Line makes a local VoIP call to Line 1. If Line 1 is busy, it stops. After a given number of rings, the VoIP gateway picks up the call.
Symmetric RTP (SPA3102 and SPA8800)
The
Symmetric RTP
port of the inbound RTP packets. This facilitates NAT traversal. See “SIP
Parameters section,” on page 127.
parameter is used to send audio RTP to the source IP and
Page 100
Configuring the PSTN (FXO) Gateway on the SPA3102
Call Scenarios
Call Progress Tones (SPA3102 and SPA8800)
The SPA3102 and the SPA8800 have configurable call progress tones. Call progress tones are generated locally on the ATA, so an end user is advised of status (such as ringback). Parameters for each type of tone (for instance a dial tone played back to an end user) may include the following specifications:
• Number of frequency components
• Frequency and amplitude of each component
• Cadence information
When one VoIP account is shared between the FXS and PSTN Lines, the following parameters are recommended to be set. For more information, see the Regional page in the “ATA Voice Field Reference,” on page 115.
6
Call Scenarios
Call Progress Tone Description
VoIP PIN Tone This tone is played to prompt a VoIP caller to enter a PIN number.
PSTN PIN Tone This tone is played to prompt a PSTN caller to enter a PIN
number.
Outside Dial Tone During two-stage PSTN-gateway dialing and with a dial plan
assigned, the ATA device collects digits from the VoIP caller and processes the number using the dial plan. The ATA device plays
Outside Dial Tone
the PSTN number. This tone should be specified to sound different from the PSTN dial tone.
This section describes some typical scenarios where the ATA device can be applied. Some terms are introduced in the first few sections and reused in later sections. This section includes the following topics:
to prompt the VoIP caller to enter the
• “PSTN to VoIP Call with and Without Ring-Thru” section on page 101
• “VoIP to PSTN Call With and Without Authentication” section on page 101
• “Call Forwarding to PSTN Gateway (SPA3102 and SPA8800)” section on
page 103
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